pc/peer_connection_integrationtest.cc - src - Git at Google

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
 #if !defined(THREAD_SANITIZER)

 #include <stdio.h>

 #include <functional>
 #include <list>
 #include <map>
 #include <memory>
 #include <utility>
 #include <vector>

 #include "absl/algorithm/container.h"
 #include "absl/memory/memory.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/media_stream_interface.h"
 #include "api/peer_connection_interface.h"
 #include "api/peer_connection_proxy.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/test/loopback_media_transport.h"
 #include "api/uma_metrics.h"
 #include "api/video_codecs/builtin_video_decoder_factory.h"
 #include "api/video_codecs/builtin_video_encoder_factory.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "call/call.h"
 #include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
 #include "logging/rtc_event_log/rtc_event_log_factory.h"
 #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
 #include "media/engine/fake_webrtc_video_engine.h"
 #include "media/engine/webrtc_media_engine.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "p2p/base/mock_async_resolver.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/port_interface.h"
 #include "p2p/base/test_stun_server.h"
 #include "p2p/base/test_turn_customizer.h"
 #include "p2p/base/test_turn_server.h"
 #include "p2p/client/basic_port_allocator.h"
 #include "pc/dtmf_sender.h"
 #include "pc/local_audio_source.h"
 #include "pc/media_session.h"
 #include "pc/peer_connection.h"
 #include "pc/peer_connection_factory.h"
 #include "pc/rtp_media_utils.h"
 #include "pc/session_description.h"
 #include "pc/test/fake_audio_capture_module.h"
 #include "pc/test/fake_periodic_video_track_source.h"
 #include "pc/test/fake_rtc_certificate_generator.h"
 #include "pc/test/fake_video_track_renderer.h"
 #include "pc/test/mock_peer_connection_observers.h"
 #include "rtc_base/fake_clock.h"
 #include "rtc_base/fake_mdns_responder.h"
 #include "rtc_base/fake_network.h"
 #include "rtc_base/firewall_socket_server.h"
 #include "rtc_base/gunit.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/test_certificate_verifier.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/virtual_socket_server.h"
 #include "system_wrappers/include/metrics.h"
 #include "test/field_trial.h"
 #include "test/gmock.h"

 namespace webrtc {
 namespace {

 using ::cricket::ContentInfo;
 using ::cricket::StreamParams;
 using ::rtc::SocketAddress;
 using ::testing::_;
 using ::testing::Combine;
 using ::testing::Contains;
 using ::testing::DoAll;
 using ::testing::ElementsAre;
 using ::testing::NiceMock;
 using ::testing::Return;
 using ::testing::SetArgPointee;
 using ::testing::UnorderedElementsAreArray;
 using ::testing::Values;
 using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;

 static const int kDefaultTimeout = 10000;
 static const int kMaxWaitForStatsMs = 3000;
 static const int kMaxWaitForActivationMs = 5000;
 static const int kMaxWaitForFramesMs = 10000;
 // Default number of audio/video frames to wait for before considering a test
 // successful.
 static const int kDefaultExpectedAudioFrameCount = 3;
 static const int kDefaultExpectedVideoFrameCount = 3;

 static const char kDataChannelLabel[] = "data_channel";

 // SRTP cipher name negotiated by the tests. This must be updated if the
 // default changes.
 static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_80;
 static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;

 static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0);

 // Helper function for constructing offer/answer options to initiate an ICE
 // restart.
 PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() {
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.ice_restart = true;
   return options;
 }

 // Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic"
 // attribute from received SDP, simulating a legacy endpoint.
 void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
   for (ContentInfo& content : desc->contents()) {
     content.media_description()->mutable_streams().clear();
   }
   desc->set_msid_supported(false);
   desc->set_msid_signaling(0);
 }

 // Removes all stream information besides the stream ids, simulating an
 // endpoint that only signals a=msid lines to convey stream_ids.
 void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
   for (ContentInfo& content : desc->contents()) {
     std::string track_id;
     std::vector<std::string> stream_ids;
     if (!content.media_description()->streams().empty()) {
       const StreamParams& first_stream =
           content.media_description()->streams()[0];
       track_id = first_stream.id;
       stream_ids = first_stream.stream_ids();
     }
     content.media_description()->mutable_streams().clear();
     StreamParams new_stream;
     new_stream.id = track_id;
     new_stream.set_stream_ids(stream_ids);
     content.media_description()->AddStream(new_stream);
   }
 }

 int FindFirstMediaStatsIndexByKind(
     const std::string& kind,
     const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
         media_stats_vec) {
   for (size_t i = 0; i < media_stats_vec.size(); i++) {
     if (media_stats_vec[i]->kind.ValueToString() == kind) {
       return i;
     }
   }
   return -1;
 }

 class SignalingMessageReceiver {
  public:
   virtual void ReceiveSdpMessage(SdpType type, const std::string& msg) = 0;
   virtual void ReceiveIceMessage(const std::string& sdp_mid,
                                  int sdp_mline_index,
                                  const std::string& msg) = 0;

  protected:
   SignalingMessageReceiver() {}
   virtual ~SignalingMessageReceiver() {}
 };

 class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
  public:
   explicit MockRtpReceiverObserver(cricket::MediaType media_type)
       : expected_media_type_(media_type) {}

   void OnFirstPacketReceived(cricket::MediaType media_type) override {
     ASSERT_EQ(expected_media_type_, media_type);
     first_packet_received_ = true;
   }

   bool first_packet_received() const { return first_packet_received_; }

   virtual ~MockRtpReceiverObserver() {}

  private:
   bool first_packet_received_ = false;
   cricket::MediaType expected_media_type_;
 };

 // Helper class that wraps a peer connection, observes it, and can accept
 // signaling messages from another wrapper.
 //
 // Uses a fake network, fake A/V capture, and optionally fake
 // encoders/decoders, though they aren't used by default since they don't
 // advertise support of any codecs.
 // TODO(steveanton): See how this could become a subclass of
 // PeerConnectionWrapper defined in peerconnectionwrapper.h.
 class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
                               public SignalingMessageReceiver {
  public:
   // Different factory methods for convenience.
   // TODO(deadbeef): Could use the pattern of:
   //
   // PeerConnectionWrapper =
   //     WrapperBuilder.WithConfig(...).WithOptions(...).build();
   //
   // To reduce some code duplication.
   static PeerConnectionWrapper* CreateWithDtlsIdentityStore(
       const std::string& debug_name,
       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
       rtc::Thread* network_thread,
       rtc::Thread* worker_thread) {
     PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
     webrtc::PeerConnectionDependencies dependencies(nullptr);
     dependencies.cert_generator = std::move(cert_generator);
     if (!client->Init(nullptr, nullptr, std::move(dependencies), network_thread,
                       worker_thread, nullptr,
                       /*media_transport_factory=*/nullptr)) {
       delete client;
       return nullptr;
     }
     return client;
   }

   webrtc::PeerConnectionFactoryInterface* pc_factory() const {
     return peer_connection_factory_.get();
   }

   webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }

   // If a signaling message receiver is set (via ConnectFakeSignaling), this
   // will set the whole offer/answer exchange in motion. Just need to wait for
   // the signaling state to reach "stable".
   void CreateAndSetAndSignalOffer() {
     auto offer = CreateOffer();
     ASSERT_NE(nullptr, offer);
     EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer)));
   }

   // Sets the options to be used when CreateAndSetAndSignalOffer is called, or
   // when a remote offer is received (via fake signaling) and an answer is
   // generated. By default, uses default options.
   void SetOfferAnswerOptions(
       const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
     offer_answer_options_ = options;
   }

   // Set a callback to be invoked when SDP is received via the fake signaling
   // channel, which provides an opportunity to munge (modify) the SDP. This is
   // used to test SDP being applied that a PeerConnection would normally not
   // generate, but a non-JSEP endpoint might.
   void SetReceivedSdpMunger(
       std::function<void(cricket::SessionDescription*)> munger) {
     received_sdp_munger_ = std::move(munger);
   }

   // Similar to the above, but this is run on SDP immediately after it's
   // generated.
   void SetGeneratedSdpMunger(
       std::function<void(cricket::SessionDescription*)> munger) {
     generated_sdp_munger_ = std::move(munger);
   }

   // Set a callback to be invoked when a remote offer is received via the fake
   // signaling channel. This provides an opportunity to change the
   // PeerConnection state before an answer is created and sent to the caller.
   void SetRemoteOfferHandler(std::function<void()> handler) {
     remote_offer_handler_ = std::move(handler);
   }

   void SetRemoteAsyncResolver(rtc::MockAsyncResolver* resolver) {
     remote_async_resolver_ = resolver;
   }

   // Every ICE connection state in order that has been seen by the observer.
   std::vector<PeerConnectionInterface::IceConnectionState>
   ice_connection_state_history() const {
     return ice_connection_state_history_;
   }
   void clear_ice_connection_state_history() {
     ice_connection_state_history_.clear();
   }

   // Every standardized ICE connection state in order that has been seen by the
   // observer.
   std::vector<PeerConnectionInterface::IceConnectionState>
   standardized_ice_connection_state_history() const {
     return standardized_ice_connection_state_history_;
   }

   // Every PeerConnection state in order that has been seen by the observer.
   std::vector<PeerConnectionInterface::PeerConnectionState>
   peer_connection_state_history() const {
     return peer_connection_state_history_;
   }

   // Every ICE gathering state in order that has been seen by the observer.
   std::vector<PeerConnectionInterface::IceGatheringState>
   ice_gathering_state_history() const {
     return ice_gathering_state_history_;
   }

   void AddAudioVideoTracks() {
     AddAudioTrack();
     AddVideoTrack();
   }

   rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() {
     return AddTrack(CreateLocalAudioTrack());
   }

   rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() {
     return AddTrack(CreateLocalVideoTrack());
   }

   rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
     cricket::AudioOptions options;
     // Disable highpass filter so that we can get all the test audio frames.
     options.highpass_filter = false;
     rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
         peer_connection_factory_->CreateAudioSource(options);
     // TODO(perkj): Test audio source when it is implemented. Currently audio
     // always use the default input.
     return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
                                                       source);
   }

   rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
     webrtc::FakePeriodicVideoSource::Config config;
     config.timestamp_offset_ms = rtc::TimeMillis();
     return CreateLocalVideoTrackInternal(config);
   }

   rtc::scoped_refptr<webrtc::VideoTrackInterface>
   CreateLocalVideoTrackWithConfig(
       webrtc::FakePeriodicVideoSource::Config config) {
     return CreateLocalVideoTrackInternal(config);
   }

   rtc::scoped_refptr<webrtc::VideoTrackInterface>
   CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
     webrtc::FakePeriodicVideoSource::Config config;
     config.rotation = rotation;
     config.timestamp_offset_ms = rtc::TimeMillis();
     return CreateLocalVideoTrackInternal(config);
   }

   rtc::scoped_refptr<RtpSenderInterface> AddTrack(
       rtc::scoped_refptr<MediaStreamTrackInterface> track,
       const std::vector<std::string>& stream_ids = {}) {
     auto result = pc()->AddTrack(track, stream_ids);
     EXPECT_EQ(RTCErrorType::NONE, result.error().type());
     return result.MoveValue();
   }

   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(
       cricket::MediaType media_type) {
     std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
     for (const auto& receiver : pc()->GetReceivers()) {
       if (receiver->media_type() == media_type) {
         receivers.push_back(receiver);
       }
     }
     return receivers;
   }

   rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType(
       cricket::MediaType media_type) {
     for (auto transceiver : pc()->GetTransceivers()) {
       if (transceiver->receiver()->media_type() == media_type) {
         return transceiver;
       }
     }
     return nullptr;
   }

   bool SignalingStateStable() {
     return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
   }

   void CreateDataChannel() { CreateDataChannel(nullptr); }

   void CreateDataChannel(const webrtc::DataChannelInit* init) {
     CreateDataChannel(kDataChannelLabel, init);
   }

   void CreateDataChannel(const std::string& label,
                          const webrtc::DataChannelInit* init) {
     data_channel_ = pc()->CreateDataChannel(label, init);
     ASSERT_TRUE(data_channel_.get() != nullptr);
     data_observer_.reset(new MockDataChannelObserver(data_channel_));
   }

   DataChannelInterface* data_channel() { return data_channel_; }
   const MockDataChannelObserver* data_observer() const {
     return data_observer_.get();
   }

   int audio_frames_received() const {
     return fake_audio_capture_module_->frames_received();
   }

   // Takes minimum of video frames received for each track.
   //
   // Can be used like:
   // EXPECT_GE(expected_frames, min_video_frames_received_per_track());
   //
   // To ensure that all video tracks received at least a certain number of
   // frames.
   int min_video_frames_received_per_track() const {
     int min_frames = INT_MAX;
     if (fake_video_renderers_.empty()) {
       return 0;
     }

     for (const auto& pair : fake_video_renderers_) {
       min_frames = std::min(min_frames, pair.second->num_rendered_frames());
     }
     return min_frames;
   }

   // Returns a MockStatsObserver in a state after stats gathering finished,
   // which can be used to access the gathered stats.
   rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
       webrtc::MediaStreamTrackInterface* track) {
     rtc::scoped_refptr<MockStatsObserver> observer(
         new rtc::RefCountedObject<MockStatsObserver>());
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
     return observer;
   }

   // Version that doesn't take a track "filter", and gathers all stats.
   rtc::scoped_refptr<MockStatsObserver> OldGetStats() {
     return OldGetStatsForTrack(nullptr);
   }

   // Synchronously gets stats and returns them. If it times out, fails the test
   // and returns null.
   rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
     rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
         new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
     peer_connection_->GetStats(callback);
     EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
     return callback->report();
   }

   int rendered_width() {
     EXPECT_FALSE(fake_video_renderers_.empty());
     return fake_video_renderers_.empty()
                ? 0
                : fake_video_renderers_.begin()->second->width();
   }

   int rendered_height() {
     EXPECT_FALSE(fake_video_renderers_.empty());
     return fake_video_renderers_.empty()
                ? 0
                : fake_video_renderers_.begin()->second->height();
   }

   double rendered_aspect_ratio() {
     if (rendered_height() == 0) {
       return 0.0;
     }
     return static_cast<double>(rendered_width()) / rendered_height();
   }

   webrtc::VideoRotation rendered_rotation() {
     EXPECT_FALSE(fake_video_renderers_.empty());
     return fake_video_renderers_.empty()
                ? webrtc::kVideoRotation_0
                : fake_video_renderers_.begin()->second->rotation();
   }

   int local_rendered_width() {
     return local_video_renderer_ ? local_video_renderer_->width() : 0;
   }

   int local_rendered_height() {
     return local_video_renderer_ ? local_video_renderer_->height() : 0;
   }

   double local_rendered_aspect_ratio() {
     if (local_rendered_height() == 0) {
       return 0.0;
     }
     return static_cast<double>(local_rendered_width()) /
            local_rendered_height();
   }

   size_t number_of_remote_streams() {
     if (!pc()) {
       return 0;
     }
     return pc()->remote_streams()->count();
   }

   StreamCollectionInterface* remote_streams() const {
     if (!pc()) {
       ADD_FAILURE();
       return nullptr;
     }
     return pc()->remote_streams();
   }

   StreamCollectionInterface* local_streams() {
     if (!pc()) {
       ADD_FAILURE();
       return nullptr;
     }
     return pc()->local_streams();
   }

   webrtc::PeerConnectionInterface::SignalingState signaling_state() {
     return pc()->signaling_state();
   }

   webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
     return pc()->ice_connection_state();
   }

   webrtc::PeerConnectionInterface::IceConnectionState
   standardized_ice_connection_state() {
     return pc()->standardized_ice_connection_state();
   }

   webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
     return pc()->ice_gathering_state();
   }

   // Returns a MockRtpReceiverObserver for each RtpReceiver returned by
   // GetReceivers. They're updated automatically when a remote offer/answer
   // from the fake signaling channel is applied, or when
   // ResetRtpReceiverObservers below is called.
   const std::vector<std::unique_ptr<MockRtpReceiverObserver>>&
   rtp_receiver_observers() {
     return rtp_receiver_observers_;
   }

   void ResetRtpReceiverObservers() {
     rtp_receiver_observers_.clear();
     for (const rtc::scoped_refptr<RtpReceiverInterface>& receiver :
          pc()->GetReceivers()) {
       std::unique_ptr<MockRtpReceiverObserver> observer(
           new MockRtpReceiverObserver(receiver->media_type()));
       receiver->SetObserver(observer.get());
       rtp_receiver_observers_.push_back(std::move(observer));
     }
   }

   rtc::FakeNetworkManager* network_manager() const {
     return fake_network_manager_.get();
   }
   cricket::PortAllocator* port_allocator() const { return port_allocator_; }

   webrtc::FakeRtcEventLogFactory* event_log_factory() const {
     return event_log_factory_;
   }

   const cricket::Candidate& last_candidate_gathered() const {
     return last_candidate_gathered_;
   }
   const cricket::IceCandidateErrorEvent& error_event() const {
     return error_event_;
   }

   // Sets the mDNS responder for the owned fake network manager and keeps a
   // reference to the responder.
   void SetMdnsResponder(
       std::unique_ptr<webrtc::FakeMdnsResponder> mdns_responder) {
     RTC_DCHECK(mdns_responder != nullptr);
     mdns_responder_ = mdns_responder.get();
     network_manager()->set_mdns_responder(std::move(mdns_responder));
   }

  private:
   explicit PeerConnectionWrapper(const std::string& debug_name)
       : debug_name_(debug_name) {}

   bool Init(
       const PeerConnectionFactory::Options* options,
       const PeerConnectionInterface::RTCConfiguration* config,
       webrtc::PeerConnectionDependencies dependencies,
       rtc::Thread* network_thread,
       rtc::Thread* worker_thread,
       std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
       std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
     // There's an error in this test code if Init ends up being called twice.
     RTC_DCHECK(!peer_connection_);
     RTC_DCHECK(!peer_connection_factory_);

     fake_network_manager_.reset(new rtc::FakeNetworkManager());
     fake_network_manager_->AddInterface(kDefaultLocalAddress);

     std::unique_ptr<cricket::PortAllocator> port_allocator(
         new cricket::BasicPortAllocator(fake_network_manager_.get()));
     port_allocator_ = port_allocator.get();
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
     if (!fake_audio_capture_module_) {
       return false;
     }
     rtc::Thread* const signaling_thread = rtc::Thread::Current();

     webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
     pc_factory_dependencies.network_thread = network_thread;
     pc_factory_dependencies.worker_thread = worker_thread;
     pc_factory_dependencies.signaling_thread = signaling_thread;
     pc_factory_dependencies.media_engine =
         cricket::WebRtcMediaEngineFactory::Create(
             rtc::scoped_refptr<webrtc::AudioDeviceModule>(
                 fake_audio_capture_module_),
             webrtc::CreateBuiltinAudioEncoderFactory(),
             webrtc::CreateBuiltinAudioDecoderFactory(),
             webrtc::CreateBuiltinVideoEncoderFactory(),
             webrtc::CreateBuiltinVideoDecoderFactory(), nullptr,
             webrtc::AudioProcessingBuilder().Create());
     pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
     if (event_log_factory) {
       event_log_factory_ = event_log_factory.get();
       pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
     } else {
       pc_factory_dependencies.event_log_factory =
           webrtc::CreateRtcEventLogFactory();
     }
     if (media_transport_factory) {
       pc_factory_dependencies.media_transport_factory =
           std::move(media_transport_factory);
     }
     peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
         std::move(pc_factory_dependencies));

     if (!peer_connection_factory_) {
       return false;
     }
     if (options) {
       peer_connection_factory_->SetOptions(*options);
     }
     if (config) {
       sdp_semantics_ = config->sdp_semantics;
     }

     dependencies.allocator = std::move(port_allocator);
     peer_connection_ = CreatePeerConnection(config, std::move(dependencies));
     return peer_connection_.get() != nullptr;
   }

   rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration* config,
       webrtc::PeerConnectionDependencies dependencies) {
     PeerConnectionInterface::RTCConfiguration modified_config;
     // If |config| is null, this will result in a default configuration being
     // used.
     if (config) {
       modified_config = *config;
     }
     // Disable resolution adaptation; we don't want it interfering with the
     // test results.
     // TODO(deadbeef): Do something more robust. Since we're testing for aspect
     // ratios and not specific resolutions, is this even necessary?
     modified_config.set_cpu_adaptation(false);

     dependencies.observer = this;
     return peer_connection_factory_->CreatePeerConnection(
         modified_config, std::move(dependencies));
   }

   void set_signaling_message_receiver(
       SignalingMessageReceiver* signaling_message_receiver) {
     signaling_message_receiver_ = signaling_message_receiver;
   }

   void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; }

   void set_signal_ice_candidates(bool signal) {
     signal_ice_candidates_ = signal;
   }

   rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
       webrtc::FakePeriodicVideoSource::Config config) {
     // Set max frame rate to 10fps to reduce the risk of test flakiness.
     // TODO(deadbeef): Do something more robust.
     config.frame_interval_ms = 100;

     video_track_sources_.emplace_back(
         new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
             config, false /* remote */));
     rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
         peer_connection_factory_->CreateVideoTrack(
             rtc::CreateRandomUuid(), video_track_sources_.back()));
     if (!local_video_renderer_) {
       local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
     }
     return track;
   }

   void HandleIncomingOffer(const std::string& msg) {
     RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer";
     std::unique_ptr<SessionDescriptionInterface> desc =
         webrtc::CreateSessionDescription(SdpType::kOffer, msg);
     if (received_sdp_munger_) {
       received_sdp_munger_(desc->description());
     }

     EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
     // Setting a remote description may have changed the number of receivers,
     // so reset the receiver observers.
     ResetRtpReceiverObservers();
     if (remote_offer_handler_) {
       remote_offer_handler_();
     }
     auto answer = CreateAnswer();
     ASSERT_NE(nullptr, answer);
     EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer)));
   }

   void HandleIncomingAnswer(const std::string& msg) {
     RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer";
     std::unique_ptr<SessionDescriptionInterface> desc =
         webrtc::CreateSessionDescription(SdpType::kAnswer, msg);
     if (received_sdp_munger_) {
       received_sdp_munger_(desc->description());
     }

     EXPECT_TRUE(SetRemoteDescription(std::move(desc)));
     // Set the RtpReceiverObserver after receivers are created.
     ResetRtpReceiverObservers();
   }

   // Returns null on failure.
   std::unique_ptr<SessionDescriptionInterface> CreateOffer() {
     rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
         new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
     pc()->CreateOffer(observer, offer_answer_options_);
     return WaitForDescriptionFromObserver(observer);
   }

   // Returns null on failure.
   std::unique_ptr<SessionDescriptionInterface> CreateAnswer() {
     rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
         new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
     pc()->CreateAnswer(observer, offer_answer_options_);
     return WaitForDescriptionFromObserver(observer);
   }

   std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver(
       MockCreateSessionDescriptionObserver* observer) {
     EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
     if (!observer->result()) {
       return nullptr;
     }
     auto description = observer->MoveDescription();
     if (generated_sdp_munger_) {
       generated_sdp_munger_(description->description());
     }
     return description;
   }

   // Setting the local description and sending the SDP message over the fake
   // signaling channel are combined into the same method because the SDP
   // message needs to be sent as soon as SetLocalDescription finishes, without
   // waiting for the observer to be called. This ensures that ICE candidates
   // don't outrace the description.
   bool SetLocalDescriptionAndSendSdpMessage(
       std::unique_ptr<SessionDescriptionInterface> desc) {
     rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
         new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
     RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage";
     SdpType type = desc->GetType();
     std::string sdp;
     EXPECT_TRUE(desc->ToString(&sdp));
     pc()->SetLocalDescription(observer, desc.release());
     if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
       RemoveUnusedVideoRenderers();
     }
     // As mentioned above, we need to send the message immediately after
     // SetLocalDescription.
     SendSdpMessage(type, sdp);
     EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
     return true;
   }

   bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) {
     rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
         new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
     RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription";
     pc()->SetRemoteDescription(observer, desc.release());
     if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
       RemoveUnusedVideoRenderers();
     }
     EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
     return observer->result();
   }

   // This is a work around to remove unused fake_video_renderers from
   // transceivers that have either stopped or are no longer receiving.
   void RemoveUnusedVideoRenderers() {
     auto transceivers = pc()->GetTransceivers();
     for (auto& transceiver : transceivers) {
       if (transceiver->receiver()->media_type() != cricket::MEDIA_TYPE_VIDEO) {
         continue;
       }
       // Remove fake video renderers from any stopped transceivers.
       if (transceiver->stopped()) {
         auto it =
             fake_video_renderers_.find(transceiver->receiver()->track()->id());
         if (it != fake_video_renderers_.end()) {
           fake_video_renderers_.erase(it);
         }
       }
       // Remove fake video renderers from any transceivers that are no longer
       // receiving.
       if ((transceiver->current_direction() &&
            !webrtc::RtpTransceiverDirectionHasRecv(
                *transceiver->current_direction()))) {
         auto it =
             fake_video_renderers_.find(transceiver->receiver()->track()->id());
         if (it != fake_video_renderers_.end()) {
           fake_video_renderers_.erase(it);
         }
       }
     }
   }

   // Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by
   // default).
   void SendSdpMessage(SdpType type, const std::string& msg) {
     if (signaling_delay_ms_ == 0) {
       RelaySdpMessageIfReceiverExists(type, msg);
     } else {
       invoker_.AsyncInvokeDelayed<void>(
           RTC_FROM_HERE, rtc::Thread::Current(),
           rtc::Bind(&PeerConnectionWrapper::RelaySdpMessageIfReceiverExists,
                     this, type, msg),
           signaling_delay_ms_);
     }
   }

   void RelaySdpMessageIfReceiverExists(SdpType type, const std::string& msg) {
     if (signaling_message_receiver_) {
       signaling_message_receiver_->ReceiveSdpMessage(type, msg);
     }
   }

   // Simulate trickling an ICE candidate with delay |signaling_delay_ms_| (0 by
   // default).
   void SendIceMessage(const std::string& sdp_mid,
                       int sdp_mline_index,
                       const std::string& msg) {
     if (signaling_delay_ms_ == 0) {
       RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index, msg);
     } else {
       invoker_.AsyncInvokeDelayed<void>(
           RTC_FROM_HERE, rtc::Thread::Current(),
           rtc::Bind(&PeerConnectionWrapper::RelayIceMessageIfReceiverExists,
                     this, sdp_mid, sdp_mline_index, msg),
           signaling_delay_ms_);
     }
   }

   void RelayIceMessageIfReceiverExists(const std::string& sdp_mid,
                                        int sdp_mline_index,
                                        const std::string& msg) {
     if (signaling_message_receiver_) {
       signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
                                                      msg);
     }
   }

   // SignalingMessageReceiver callbacks.
   void ReceiveSdpMessage(SdpType type, const std::string& msg) override {
     if (type == SdpType::kOffer) {
       HandleIncomingOffer(msg);
     } else {
       HandleIncomingAnswer(msg);
     }
   }

   void ReceiveIceMessage(const std::string& sdp_mid,
                          int sdp_mline_index,
                          const std::string& msg) override {
     RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
     std::unique_ptr<webrtc::IceCandidateInterface> candidate(
         webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
     EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
   }

   // PeerConnectionObserver callbacks.
   void OnSignalingChange(
       webrtc::PeerConnectionInterface::SignalingState new_state) override {
     EXPECT_EQ(pc()->signaling_state(), new_state);
   }
   void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
                   const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
                       streams) override {
     if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
       rtc::scoped_refptr<VideoTrackInterface> video_track(
           static_cast<VideoTrackInterface*>(receiver->track().get()));
       ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
                   fake_video_renderers_.end());
       fake_video_renderers_[video_track->id()] =
           absl::make_unique<FakeVideoTrackRenderer>(video_track);
     }
   }
   void OnRemoveTrack(
       rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
     if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
       auto it = fake_video_renderers_.find(receiver->track()->id());
       RTC_DCHECK(it != fake_video_renderers_.end());
       fake_video_renderers_.erase(it);
     }
   }
   void OnRenegotiationNeeded() override {}
   void OnIceConnectionChange(
       webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
     EXPECT_EQ(pc()->ice_connection_state(), new_state);
     ice_connection_state_history_.push_back(new_state);
   }
   void OnStandardizedIceConnectionChange(
       webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
     standardized_ice_connection_state_history_.push_back(new_state);
   }
   void OnConnectionChange(
       webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
     peer_connection_state_history_.push_back(new_state);
   }

   void OnIceGatheringChange(
       webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
     EXPECT_EQ(pc()->ice_gathering_state(), new_state);
     ice_gathering_state_history_.push_back(new_state);
   }
   void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
     RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate";

     if (remote_async_resolver_) {
       const auto& local_candidate = candidate->candidate();
       if (local_candidate.address().IsUnresolvedIP()) {
         RTC_DCHECK(local_candidate.type() == cricket::LOCAL_PORT_TYPE);
         rtc::SocketAddress resolved_addr(local_candidate.address());
         const auto resolved_ip = mdns_responder_->GetMappedAddressForName(
             local_candidate.address().hostname());
         RTC_DCHECK(!resolved_ip.IsNil());
         resolved_addr.SetResolvedIP(resolved_ip);
         EXPECT_CALL(*remote_async_resolver_, GetResolvedAddress(_, _))
             .WillOnce(DoAll(SetArgPointee<1>(resolved_addr), Return(true)));
         EXPECT_CALL(*remote_async_resolver_, Destroy(_));
       }
     }

     std::string ice_sdp;
     EXPECT_TRUE(candidate->ToString(&ice_sdp));
     if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) {
       // Remote party may be deleted.
       return;
     }
     SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
     last_candidate_gathered_ = candidate->candidate();
   }
   void OnIceCandidateError(const std::string& host_candidate,
                            const std::string& url,
                            int error_code,
                            const std::string& error_text) override {
     error_event_ = cricket::IceCandidateErrorEvent(host_candidate, url,
                                                    error_code, error_text);
   }
   void OnDataChannel(
       rtc::scoped_refptr<DataChannelInterface> data_channel) override {
     RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel";
     data_channel_ = data_channel;
     data_observer_.reset(new MockDataChannelObserver(data_channel));
   }

   std::string debug_name_;

   std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
   // Reference to the mDNS responder owned by |fake_network_manager_| after set.
   webrtc::FakeMdnsResponder* mdns_responder_ = nullptr;

   rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
       peer_connection_factory_;

   cricket::PortAllocator* port_allocator_;
   // Needed to keep track of number of frames sent.
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
   // Needed to keep track of number of frames received.
   std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
       fake_video_renderers_;
   // Needed to ensure frames aren't received for removed tracks.
   std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
       removed_fake_video_renderers_;

   // For remote peer communication.
   SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
   int signaling_delay_ms_ = 0;
   bool signal_ice_candidates_ = true;
   cricket::Candidate last_candidate_gathered_;
   cricket::IceCandidateErrorEvent error_event_;

   // Store references to the video sources we've created, so that we can stop
   // them, if required.
   std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
       video_track_sources_;
   // |local_video_renderer_| attached to the first created local video track.
   std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;

   SdpSemantics sdp_semantics_;
   PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
   std::function<void(cricket::SessionDescription*)> received_sdp_munger_;
   std::function<void(cricket::SessionDescription*)> generated_sdp_munger_;
   std::function<void()> remote_offer_handler_;
   rtc::MockAsyncResolver* remote_async_resolver_ = nullptr;
   rtc::scoped_refptr<DataChannelInterface> data_channel_;
   std::unique_ptr<MockDataChannelObserver> data_observer_;

   std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;

   std::vector<PeerConnectionInterface::IceConnectionState>
       ice_connection_state_history_;
   std::vector<PeerConnectionInterface::IceConnectionState>
       standardized_ice_connection_state_history_;
   std::vector<PeerConnectionInterface::PeerConnectionState>
       peer_connection_state_history_;
   std::vector<PeerConnectionInterface::IceGatheringState>
       ice_gathering_state_history_;

   webrtc::FakeRtcEventLogFactory* event_log_factory_;

   rtc::AsyncInvoker invoker_;

   friend class PeerConnectionIntegrationBaseTest;
 };

 class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
  public:
   virtual ~MockRtcEventLogOutput() = default;
   MOCK_CONST_METHOD0(IsActive, bool());
   MOCK_METHOD1(Write, bool(const std::string&));
 };

 // This helper object is used for both specifying how many audio/video frames
 // are expected to be received for a caller/callee. It provides helper functions
 // to specify these expectations. The object initially starts in a state of no
 // expectations.
 class MediaExpectations {
  public:
   enum ExpectFrames {
     kExpectSomeFrames,
     kExpectNoFrames,
     kNoExpectation,
   };

   void ExpectBidirectionalAudioAndVideo() {
     ExpectBidirectionalAudio();
     ExpectBidirectionalVideo();
   }

   void ExpectBidirectionalAudio() {
     CallerExpectsSomeAudio();
     CalleeExpectsSomeAudio();
   }

   void ExpectNoAudio() {
     CallerExpectsNoAudio();
     CalleeExpectsNoAudio();
   }

   void ExpectBidirectionalVideo() {
     CallerExpectsSomeVideo();
     CalleeExpectsSomeVideo();
   }

   void ExpectNoVideo() {
     CallerExpectsNoVideo();
     CalleeExpectsNoVideo();
   }

   void CallerExpectsSomeAudioAndVideo() {
     CallerExpectsSomeAudio();
     CallerExpectsSomeVideo();
   }

   void CalleeExpectsSomeAudioAndVideo() {
     CalleeExpectsSomeAudio();
     CalleeExpectsSomeVideo();
   }

   // Caller's audio functions.
   void CallerExpectsSomeAudio(
       int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
     caller_audio_expectation_ = kExpectSomeFrames;
     caller_audio_frames_expected_ = expected_audio_frames;
   }

   void CallerExpectsNoAudio() {
     caller_audio_expectation_ = kExpectNoFrames;
     caller_audio_frames_expected_ = 0;
   }

   // Caller's video functions.
   void CallerExpectsSomeVideo(
       int expected_video_frames = kDefaultExpectedVideoFrameCount) {
     caller_video_expectation_ = kExpectSomeFrames;
     caller_video_frames_expected_ = expected_video_frames;
   }

   void CallerExpectsNoVideo() {
     caller_video_expectation_ = kExpectNoFrames;
     caller_video_frames_expected_ = 0;
   }

   // Callee's audio functions.
   void CalleeExpectsSomeAudio(
       int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
     callee_audio_expectation_ = kExpectSomeFrames;
     callee_audio_frames_expected_ = expected_audio_frames;
   }

   void CalleeExpectsNoAudio() {
     callee_audio_expectation_ = kExpectNoFrames;
     callee_audio_frames_expected_ = 0;
   }

   // Callee's video functions.
   void CalleeExpectsSomeVideo(
       int expected_video_frames = kDefaultExpectedVideoFrameCount) {
     callee_video_expectation_ = kExpectSomeFrames;
     callee_video_frames_expected_ = expected_video_frames;
   }

   void CalleeExpectsNoVideo() {
     callee_video_expectation_ = kExpectNoFrames;
     callee_video_frames_expected_ = 0;
   }

   ExpectFrames caller_audio_expectation_ = kNoExpectation;
   ExpectFrames caller_video_expectation_ = kNoExpectation;
   ExpectFrames callee_audio_expectation_ = kNoExpectation;
   ExpectFrames callee_video_expectation_ = kNoExpectation;
   int caller_audio_frames_expected_ = 0;
   int caller_video_frames_expected_ = 0;
   int callee_audio_frames_expected_ = 0;
   int callee_video_frames_expected_ = 0;
 };

 // Tests two PeerConnections connecting to each other end-to-end, using a
 // virtual network, fake A/V capture and fake encoder/decoders. The
 // PeerConnections share the threads/socket servers, but use separate versions
 // of everything else (including "PeerConnectionFactory"s).
 class PeerConnectionIntegrationBaseTest : public ::testing::Test {
  public:
   explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics)
       : sdp_semantics_(sdp_semantics),
         ss_(new rtc::VirtualSocketServer()),
         fss_(new rtc::FirewallSocketServer(ss_.get())),
         network_thread_(new rtc::Thread(fss_.get())),
         worker_thread_(rtc::Thread::Create()),
         loopback_media_transports_(network_thread_.get()) {
     network_thread_->SetName("PCNetworkThread", this);
     worker_thread_->SetName("PCWorkerThread", this);
     RTC_CHECK(network_thread_->Start());
     RTC_CHECK(worker_thread_->Start());
     webrtc::metrics::Reset();
   }

   ~PeerConnectionIntegrationBaseTest() {
     // The PeerConnections should deleted before the TurnCustomizers.
     // A TurnPort is created with a raw pointer to a TurnCustomizer. The
     // TurnPort has the same lifetime as the PeerConnection, so it's expected
     // that the TurnCustomizer outlives the life of the PeerConnection or else
     // when Send() is called it will hit a seg fault.
     if (caller_) {
       caller_->set_signaling_message_receiver(nullptr);
       delete SetCallerPcWrapperAndReturnCurrent(nullptr);
     }
     if (callee_) {
       callee_->set_signaling_message_receiver(nullptr);
       delete SetCalleePcWrapperAndReturnCurrent(nullptr);
     }

     // If turn servers were created for the test they need to be destroyed on
     // the network thread.
     network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
       turn_servers_.clear();
       turn_customizers_.clear();
     });
   }

   bool SignalingStateStable() {
     return caller_->SignalingStateStable() && callee_->SignalingStateStable();
   }

   bool DtlsConnected() {
     // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
     // are connected. This is an important distinction. Once we have separate
     // ICE and DTLS state, this check needs to use the DTLS state.
     return (callee()->ice_connection_state() ==
                 webrtc::PeerConnectionInterface::kIceConnectionConnected ||
             callee()->ice_connection_state() ==
                 webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
            (caller()->ice_connection_state() ==
                 webrtc::PeerConnectionInterface::kIceConnectionConnected ||
             caller()->ice_connection_state() ==
                 webrtc::PeerConnectionInterface::kIceConnectionCompleted);
   }

   // When |event_log_factory| is null, the default implementation of the event
   // log factory will be used.
   std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper(
       const std::string& debug_name,
       const PeerConnectionFactory::Options* options,
       const RTCConfiguration* config,
       webrtc::PeerConnectionDependencies dependencies,
       std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
       std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
     RTCConfiguration modified_config;
     if (config) {
       modified_config = *config;
     }
     modified_config.sdp_semantics = sdp_semantics_;
     if (!dependencies.cert_generator) {
       dependencies.cert_generator =
           absl::make_unique<FakeRTCCertificateGenerator>();
     }
     std::unique_ptr<PeerConnectionWrapper> client(
         new PeerConnectionWrapper(debug_name));

     if (!client->Init(options, &modified_config, std::move(dependencies),
                       network_thread_.get(), worker_thread_.get(),
                       std::move(event_log_factory),
                       std::move(media_transport_factory))) {
       return nullptr;
     }
     return client;
   }

   std::unique_ptr<PeerConnectionWrapper>
   CreatePeerConnectionWrapperWithFakeRtcEventLog(
       const std::string& debug_name,
       const PeerConnectionFactory::Options* options,
       const RTCConfiguration* config,
       webrtc::PeerConnectionDependencies dependencies) {
     std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory(
         new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
     return CreatePeerConnectionWrapper(debug_name, options, config,
                                        std::move(dependencies),
                                        std::move(event_log_factory),
                                        /*media_transport_factory=*/nullptr);
   }

   bool CreatePeerConnectionWrappers() {
     return CreatePeerConnectionWrappersWithConfig(
         PeerConnectionInterface::RTCConfiguration(),
         PeerConnectionInterface::RTCConfiguration());
   }

   bool CreatePeerConnectionWrappersWithSdpSemantics(
       SdpSemantics caller_semantics,
       SdpSemantics callee_semantics) {
     // Can't specify the sdp_semantics in the passed-in configuration since it
     // will be overwritten by CreatePeerConnectionWrapper with whatever is
     // stored in sdp_semantics_. So get around this by modifying the instance
     // variable before calling CreatePeerConnectionWrapper for the caller and
     // callee PeerConnections.
     SdpSemantics original_semantics = sdp_semantics_;
     sdp_semantics_ = caller_semantics;
     caller_ = CreatePeerConnectionWrapper(
         "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
         nullptr, /*media_transport_factory=*/nullptr);
     sdp_semantics_ = callee_semantics;
     callee_ = CreatePeerConnectionWrapper(
         "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
         nullptr, /*media_transport_factory=*/nullptr);
     sdp_semantics_ = original_semantics;
     return caller_ && callee_;
   }

   bool CreatePeerConnectionWrappersWithConfig(
       const PeerConnectionInterface::RTCConfiguration& caller_config,
       const PeerConnectionInterface::RTCConfiguration& callee_config) {
     caller_ = CreatePeerConnectionWrapper(
         "Caller", nullptr, &caller_config,
         webrtc::PeerConnectionDependencies(nullptr), nullptr,
         /*media_transport_factory=*/nullptr);
     callee_ = CreatePeerConnectionWrapper(
         "Callee", nullptr, &callee_config,
         webrtc::PeerConnectionDependencies(nullptr), nullptr,
         /*media_transport_factory=*/nullptr);
     return caller_ && callee_;
   }

   bool CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       const PeerConnectionInterface::RTCConfiguration& caller_config,
       const PeerConnectionInterface::RTCConfiguration& callee_config,
       std::unique_ptr<webrtc::MediaTransportFactory> caller_factory,
       std::unique_ptr<webrtc::MediaTransportFactory> callee_factory) {
     caller_ =
         CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
                                     webrtc::PeerConnectionDependencies(nullptr),
                                     nullptr, std::move(caller_factory));
     callee_ =
         CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
                                     webrtc::PeerConnectionDependencies(nullptr),
                                     nullptr, std::move(callee_factory));
     return caller_ && callee_;
   }

   bool CreatePeerConnectionWrappersWithConfigAndDeps(
       const PeerConnectionInterface::RTCConfiguration& caller_config,
       webrtc::PeerConnectionDependencies caller_dependencies,
       const PeerConnectionInterface::RTCConfiguration& callee_config,
       webrtc::PeerConnectionDependencies callee_dependencies) {
     caller_ =
         CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
                                     std::move(caller_dependencies), nullptr,
                                     /*media_transport_factory=*/nullptr);
     callee_ =
         CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
                                     std::move(callee_dependencies), nullptr,
                                     /*media_transport_factory=*/nullptr);
     return caller_ && callee_;
   }

   bool CreatePeerConnectionWrappersWithOptions(
       const PeerConnectionFactory::Options& caller_options,
       const PeerConnectionFactory::Options& callee_options) {
     caller_ = CreatePeerConnectionWrapper(
         "Caller", &caller_options, nullptr,
         webrtc::PeerConnectionDependencies(nullptr), nullptr,
         /*media_transport_factory=*/nullptr);
     callee_ = CreatePeerConnectionWrapper(
         "Callee", &callee_options, nullptr,
         webrtc::PeerConnectionDependencies(nullptr), nullptr,
         /*media_transport_factory=*/nullptr);
     return caller_ && callee_;
   }

   bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
     PeerConnectionInterface::RTCConfiguration default_config;
     caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
         "Caller", nullptr, &default_config,
         webrtc::PeerConnectionDependencies(nullptr));
     callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
         "Callee", nullptr, &default_config,
         webrtc::PeerConnectionDependencies(nullptr));
     return caller_ && callee_;
   }

   std::unique_ptr<PeerConnectionWrapper>
   CreatePeerConnectionWrapperWithAlternateKey() {
     std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
         new FakeRTCCertificateGenerator());
     cert_generator->use_alternate_key();

     webrtc::PeerConnectionDependencies dependencies(nullptr);
     dependencies.cert_generator = std::move(cert_generator);
     return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
                                        std::move(dependencies), nullptr,
                                        /*media_transport_factory=*/nullptr);
   }

   cricket::TestTurnServer* CreateTurnServer(
       rtc::SocketAddress internal_address,
       rtc::SocketAddress external_address,
       cricket::ProtocolType type = cricket::ProtocolType::PROTO_UDP,
       const std::string& common_name = "test turn server") {
     rtc::Thread* thread = network_thread();
     std::unique_ptr<cricket::TestTurnServer> turn_server =
         network_thread()->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
             RTC_FROM_HERE,
             [thread, internal_address, external_address, type, common_name] {
               return absl::make_unique<cricket::TestTurnServer>(
                   thread, internal_address, external_address, type,
                   /*ignore_bad_certs=*/true, common_name);
             });
     turn_servers_.push_back(std::move(turn_server));
     // Interactions with the turn server should be done on the network thread.
     return turn_servers_.back().get();
   }

   cricket::TestTurnCustomizer* CreateTurnCustomizer() {
     std::unique_ptr<cricket::TestTurnCustomizer> turn_customizer =
         network_thread()->Invoke<std::unique_ptr<cricket::TestTurnCustomizer>>(
             RTC_FROM_HERE,
             [] { return absl::make_unique<cricket::TestTurnCustomizer>(); });
     turn_customizers_.push_back(std::move(turn_customizer));
     // Interactions with the turn customizer should be done on the network
     // thread.
     return turn_customizers_.back().get();
   }

   // Checks that the function counters for a TestTurnCustomizer are greater than
   // 0.
   void ExpectTurnCustomizerCountersIncremented(
       cricket::TestTurnCustomizer* turn_customizer) {
     unsigned int allow_channel_data_counter =
         network_thread()->Invoke<unsigned int>(
             RTC_FROM_HERE, [turn_customizer] {
               return turn_customizer->allow_channel_data_cnt_;
             });
     EXPECT_GT(allow_channel_data_counter, 0u);
     unsigned int modify_counter = network_thread()->Invoke<unsigned int>(
         RTC_FROM_HERE,
         [turn_customizer] { return turn_customizer->modify_cnt_; });
     EXPECT_GT(modify_counter, 0u);
   }

   // Once called, SDP blobs and ICE candidates will be automatically signaled
   // between PeerConnections.
   void ConnectFakeSignaling() {
     caller_->set_signaling_message_receiver(callee_.get());
     callee_->set_signaling_message_receiver(caller_.get());
   }

   // Once called, SDP blobs will be automatically signaled between
   // PeerConnections. Note that ICE candidates will not be signaled unless they
   // are in the exchanged SDP blobs.
   void ConnectFakeSignalingForSdpOnly() {
     ConnectFakeSignaling();
     SetSignalIceCandidates(false);
   }

   void SetSignalingDelayMs(int delay_ms) {
     caller_->set_signaling_delay_ms(delay_ms);
     callee_->set_signaling_delay_ms(delay_ms);
   }

   void SetSignalIceCandidates(bool signal) {
     caller_->set_signal_ice_candidates(signal);
     callee_->set_signal_ice_candidates(signal);
   }

   // Messages may get lost on the unreliable DataChannel, so we send multiple
   // times to avoid test flakiness.
   void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
                               const std::string& data,
                               int retries) {
     for (int i = 0; i < retries; ++i) {
       dc->Send(DataBuffer(data));
     }
   }

   rtc::Thread* network_thread() { return network_thread_.get(); }

   rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }

   webrtc::MediaTransportPair* loopback_media_transports() {
     return &loopback_media_transports_;
   }

   PeerConnectionWrapper* caller() { return caller_.get(); }

   // Set the |caller_| to the |wrapper| passed in and return the
   // original |caller_|.
   PeerConnectionWrapper* SetCallerPcWrapperAndReturnCurrent(
       PeerConnectionWrapper* wrapper) {
     PeerConnectionWrapper* old = caller_.release();
     caller_.reset(wrapper);
     return old;
   }

   PeerConnectionWrapper* callee() { return callee_.get(); }

   // Set the |callee_| to the |wrapper| passed in and return the
   // original |callee_|.
   PeerConnectionWrapper* SetCalleePcWrapperAndReturnCurrent(
       PeerConnectionWrapper* wrapper) {
     PeerConnectionWrapper* old = callee_.release();
     callee_.reset(wrapper);
     return old;
   }

   void SetPortAllocatorFlags(uint32_t caller_flags, uint32_t callee_flags) {
     network_thread()->Invoke<void>(
         RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
                                  caller()->port_allocator(), caller_flags));
     network_thread()->Invoke<void>(
         RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::set_flags,
                                  callee()->port_allocator(), callee_flags));
   }

   rtc::FirewallSocketServer* firewall() const { return fss_.get(); }

   // Expects the provided number of new frames to be received within
   // kMaxWaitForFramesMs. The new expected frames are specified in
   // |media_expectations|. Returns false if any of the expectations were
   // not met.
   bool ExpectNewFrames(const MediaExpectations& media_expectations) {
     // First initialize the expected frame counts based upon the current
     // frame count.
     int total_caller_audio_frames_expected = caller()->audio_frames_received();
     if (media_expectations.caller_audio_expectation_ ==
         MediaExpectations::kExpectSomeFrames) {
       total_caller_audio_frames_expected +=
           media_expectations.caller_audio_frames_expected_;
     }
     int total_caller_video_frames_expected =
         caller()->min_video_frames_received_per_track();
     if (media_expectations.caller_video_expectation_ ==
         MediaExpectations::kExpectSomeFrames) {
       total_caller_video_frames_expected +=
           media_expectations.caller_video_frames_expected_;
     }
     int total_callee_audio_frames_expected = callee()->audio_frames_received();
     if (media_expectations.callee_audio_expectation_ ==
         MediaExpectations::kExpectSomeFrames) {
       total_callee_audio_frames_expected +=
           media_expectations.callee_audio_frames_expected_;
     }
     int total_callee_video_frames_expected =
         callee()->min_video_frames_received_per_track();
     if (media_expectations.callee_video_expectation_ ==
         MediaExpectations::kExpectSomeFrames) {
       total_callee_video_frames_expected +=
           media_expectations.callee_video_frames_expected_;
     }

     // Wait for the expected frames.
     EXPECT_TRUE_WAIT(caller()->audio_frames_received() >=
                              total_caller_audio_frames_expected &&
                          caller()->min_video_frames_received_per_track() >=
                              total_caller_video_frames_expected &&
                          callee()->audio_frames_received() >=
                              total_callee_audio_frames_expected &&
                          callee()->min_video_frames_received_per_track() >=
                              total_callee_video_frames_expected,
                      kMaxWaitForFramesMs);
     bool expectations_correct =
         caller()->audio_frames_received() >=
             total_caller_audio_frames_expected &&
         caller()->min_video_frames_received_per_track() >=
             total_caller_video_frames_expected &&
         callee()->audio_frames_received() >=
             total_callee_audio_frames_expected &&
         callee()->min_video_frames_received_per_track() >=
             total_callee_video_frames_expected;

     // After the combined wait, print out a more detailed message upon
     // failure.
     EXPECT_GE(caller()->audio_frames_received(),
               total_caller_audio_frames_expected);
     EXPECT_GE(caller()->min_video_frames_received_per_track(),
               total_caller_video_frames_expected);
     EXPECT_GE(callee()->audio_frames_received(),
               total_callee_audio_frames_expected);
     EXPECT_GE(callee()->min_video_frames_received_per_track(),
               total_callee_video_frames_expected);

     // We want to make sure nothing unexpected was received.
     if (media_expectations.caller_audio_expectation_ ==
         MediaExpectations::kExpectNoFrames) {
       EXPECT_EQ(caller()->audio_frames_received(),
                 total_caller_audio_frames_expected);
       if (caller()->audio_frames_received() !=
           total_caller_audio_frames_expected) {
         expectations_correct = false;
       }
     }
     if (media_expectations.caller_video_expectation_ ==
         MediaExpectations::kExpectNoFrames) {
       EXPECT_EQ(caller()->min_video_frames_received_per_track(),
                 total_caller_video_frames_expected);
       if (caller()->min_video_frames_received_per_track() !=
           total_caller_video_frames_expected) {
         expectations_correct = false;
       }
     }
     if (media_expectations.callee_audio_expectation_ ==
         MediaExpectations::kExpectNoFrames) {
       EXPECT_EQ(callee()->audio_frames_received(),
                 total_callee_audio_frames_expected);
       if (callee()->audio_frames_received() !=
           total_callee_audio_frames_expected) {
         expectations_correct = false;
       }
     }
     if (media_expectations.callee_video_expectation_ ==
         MediaExpectations::kExpectNoFrames) {
       EXPECT_EQ(callee()->min_video_frames_received_per_track(),
                 total_callee_video_frames_expected);
       if (callee()->min_video_frames_received_per_track() !=
           total_callee_video_frames_expected) {
         expectations_correct = false;
       }
     }
     return expectations_correct;
   }

   void ClosePeerConnections() {
     caller()->pc()->Close();
     callee()->pc()->Close();
   }

   void TestNegotiatedCipherSuite(
       const PeerConnectionFactory::Options& caller_options,
       const PeerConnectionFactory::Options& callee_options,
       int expected_cipher_suite) {
     ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
                                                         callee_options));
     ConnectFakeSignaling();
     caller()->AddAudioVideoTracks();
     callee()->AddAudioVideoTracks();
     caller()->CreateAndSetAndSignalOffer();
     ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
     EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
                    caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
     // TODO(bugs.webrtc.org/9456): Fix it.
     EXPECT_EQ(1, webrtc::metrics::NumEvents(
                      "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                      expected_cipher_suite));
   }

   void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
                                          bool remote_gcm_enabled,
                                          int expected_cipher_suite) {
     PeerConnectionFactory::Options caller_options;
     caller_options.crypto_options.srtp.enable_gcm_crypto_suites =
         local_gcm_enabled;
     PeerConnectionFactory::Options callee_options;
     callee_options.crypto_options.srtp.enable_gcm_crypto_suites =
         remote_gcm_enabled;
     TestNegotiatedCipherSuite(caller_options, callee_options,
                               expected_cipher_suite);
   }

  protected:
   SdpSemantics sdp_semantics_;

  private:
   // |ss_| is used by |network_thread_| so it must be destroyed later.
   std::unique_ptr<rtc::VirtualSocketServer> ss_;
   std::unique_ptr<rtc::FirewallSocketServer> fss_;
   // |network_thread_| and |worker_thread_| are used by both
   // |caller_| and |callee_| so they must be destroyed
   // later.
   std::unique_ptr<rtc::Thread> network_thread_;
   std::unique_ptr<rtc::Thread> worker_thread_;
   // The turn servers and turn customizers should be accessed & deleted on the
   // network thread to avoid a race with the socket read/write that occurs
   // on the network thread.
   std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
   std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
   webrtc::MediaTransportPair loopback_media_transports_;
   std::unique_ptr<PeerConnectionWrapper> caller_;
   std::unique_ptr<PeerConnectionWrapper> callee_;
 };

 class PeerConnectionIntegrationTest
     : public PeerConnectionIntegrationBaseTest,
       public ::testing::WithParamInterface<SdpSemantics> {
  protected:
   PeerConnectionIntegrationTest()
       : PeerConnectionIntegrationBaseTest(GetParam()) {}
 };

 class PeerConnectionIntegrationTestPlanB
     : public PeerConnectionIntegrationBaseTest {
  protected:
   PeerConnectionIntegrationTestPlanB()
       : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {}
 };

 class PeerConnectionIntegrationTestUnifiedPlan
     : public PeerConnectionIntegrationBaseTest {
  protected:
   PeerConnectionIntegrationTestUnifiedPlan()
       : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
 };

 // Test the OnFirstPacketReceived callback from audio/video RtpReceivers.  This
 // includes testing that the callback is invoked if an observer is connected
 // after the first packet has already been received.
 TEST_P(PeerConnectionIntegrationTest,
        RtpReceiverObserverOnFirstPacketReceived) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   // Start offer/answer exchange and wait for it to complete.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Should be one receiver each for audio/video.
   EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
   EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
   // Wait for all "first packet received" callbacks to be fired.
   EXPECT_TRUE_WAIT(
       absl::c_all_of(caller()->rtp_receiver_observers(),
                      [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                        return o->first_packet_received();
                      }),
       kMaxWaitForFramesMs);
   EXPECT_TRUE_WAIT(
       absl::c_all_of(callee()->rtp_receiver_observers(),
                      [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                        return o->first_packet_received();
                      }),
       kMaxWaitForFramesMs);
   // If new observers are set after the first packet was already received, the
   // callback should still be invoked.
   caller()->ResetRtpReceiverObservers();
   callee()->ResetRtpReceiverObservers();
   EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
   EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
   EXPECT_TRUE(
       absl::c_all_of(caller()->rtp_receiver_observers(),
                      [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                        return o->first_packet_received();
                      }));
   EXPECT_TRUE(
       absl::c_all_of(callee()->rtp_receiver_observers(),
                      [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                        return o->first_packet_received();
                      }));
 }

 class DummyDtmfObserver : public DtmfSenderObserverInterface {
  public:
   DummyDtmfObserver() : completed_(false) {}

   // Implements DtmfSenderObserverInterface.
   void OnToneChange(const std::string& tone) override {
     tones_.push_back(tone);
     if (tone.empty()) {
       completed_ = true;
     }
   }

   const std::vector<std::string>& tones() const { return tones_; }
   bool completed() const { return completed_; }

  private:
   bool completed_;
   std::vector<std::string> tones_;
 };

 // Assumes |sender| already has an audio track added and the offer/answer
 // exchange is done.
 void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender,
                                   PeerConnectionWrapper* receiver) {
   // We should be able to get a DTMF sender from the local sender.
   rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender =
       sender->pc()->GetSenders().at(0)->GetDtmfSender();
   ASSERT_TRUE(dtmf_sender);
   DummyDtmfObserver observer;
   dtmf_sender->RegisterObserver(&observer);

   // Test the DtmfSender object just created.
   EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
   EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));

   EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout);
   std::vector<std::string> tones = {"1", "a", ""};
   EXPECT_EQ(tones, observer.tones());
   dtmf_sender->UnregisterObserver();
   // TODO(deadbeef): Verify the tones were actually received end-to-end.
 }

 // Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each
 // direction).
 TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Only need audio for DTMF.
   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // DTLS must finish before the DTMF sender can be used reliably.
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
   TestDtmfFromSenderToReceiver(caller(), callee());
   TestDtmfFromSenderToReceiver(callee(), caller());
 }

 // Basic end-to-end test, verifying media can be encoded/transmitted/decoded
 // between two connections, using DTLS-SRTP.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Do normal offer/answer and wait for some frames to be received in each
   // direction.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
   EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                           webrtc::kEnumCounterKeyProtocolDtls));
   EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                           webrtc::kEnumCounterKeyProtocolSdes));
 }

 // Uses SDES instead of DTLS for key agreement.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
   PeerConnectionInterface::RTCConfiguration sdes_config;
   sdes_config.enable_dtls_srtp.emplace(false);
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
   ConnectFakeSignaling();

   // Do normal offer/answer and wait for some frames to be received in each
   // direction.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
   EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                           webrtc::kEnumCounterKeyProtocolSdes));
   EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
                                           webrtc::kEnumCounterKeyProtocolDtls));
 }

 // Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
 // certificate once the DTLS handshake has finished.
 TEST_P(PeerConnectionIntegrationTest,
        GetRemoteAudioSSLCertificateReturnsExchangedCertificate) {
   auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) {
     auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
     auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
     return pc->GetRemoteAudioSSLCertificate();
   };
   auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) {
     auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
     auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
     return pc->GetRemoteAudioSSLCertChain();
   };

   auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
   auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]);

   // Configure each side with a known certificate so they can be compared later.
   PeerConnectionInterface::RTCConfiguration caller_config;
   caller_config.enable_dtls_srtp.emplace(true);
   caller_config.certificates.push_back(caller_cert);
   PeerConnectionInterface::RTCConfiguration callee_config;
   callee_config.enable_dtls_srtp.emplace(true);
   callee_config.certificates.push_back(callee_cert);
   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
   ConnectFakeSignaling();

   // When first initialized, there should not be a remote SSL certificate (and
   // calling this method should not crash).
   EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller()));
   EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee()));
   EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller()));
   EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee()));

   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);

   // Once DTLS has been connected, each side should return the other's SSL
   // certificate when calling GetRemoteAudioSSLCertificate.

   auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller());
   ASSERT_TRUE(caller_remote_cert);
   EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
             caller_remote_cert->ToPEMString());

   auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee());
   ASSERT_TRUE(callee_remote_cert);
   EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
             callee_remote_cert->ToPEMString());

   auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller());
   ASSERT_TRUE(caller_remote_cert_chain);
   ASSERT_EQ(1U, caller_remote_cert_chain->GetSize());
   auto remote_cert = &caller_remote_cert_chain->Get(0);
   EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
             remote_cert->ToPEMString());

   auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee());
   ASSERT_TRUE(callee_remote_cert_chain);
   ASSERT_EQ(1U, callee_remote_cert_chain->GetSize());
   remote_cert = &callee_remote_cert_chain->Get(0);
   EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
             remote_cert->ToPEMString());
 }

 // This test sets up a call between two parties with a source resolution of
 // 1280x720 and verifies that a 16:9 aspect ratio is received.
 TEST_P(PeerConnectionIntegrationTest,
        Send1280By720ResolutionAndReceive16To9AspectRatio) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Add video tracks with 16:9 aspect ratio, size 1280 x 720.
   webrtc::FakePeriodicVideoSource::Config config;
   config.width = 1280;
   config.height = 720;
   config.timestamp_offset_ms = rtc::TimeMillis();
   caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config));
   callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config));

   // Do normal offer/answer and wait for at least one frame to be received in
   // each direction.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                        callee()->min_video_frames_received_per_track() > 0,
                    kMaxWaitForFramesMs);

   // Check rendered aspect ratio.
   EXPECT_EQ(16.0 / 9, caller()->local_rendered_aspect_ratio());
   EXPECT_EQ(16.0 / 9, caller()->rendered_aspect_ratio());
   EXPECT_EQ(16.0 / 9, callee()->local_rendered_aspect_ratio());
   EXPECT_EQ(16.0 / 9, callee()->rendered_aspect_ratio());
 }

 // This test sets up an one-way call, with media only from caller to
 // callee.
 TEST_P(PeerConnectionIntegrationTest, OneWayMediaCall) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   media_expectations.CallerExpectsNoAudio();
   media_expectations.CallerExpectsNoVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This test sets up a audio call initially, with the callee rejecting video
 // initially. Then later the callee decides to upgrade to audio/video, and
 // initiates a new offer/answer exchange.
 TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Initially, offer an audio/video stream from the caller, but refuse to
   // send/receive video on the callee side.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioTrack();
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_video = 0;
     callee()->SetOfferAnswerOptions(options);
   } else {
     callee()->SetRemoteOfferHandler([this] {
       callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
     });
   }
   // Do offer/answer and make sure audio is still received end-to-end.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   {
     MediaExpectations media_expectations;
     media_expectations.ExpectBidirectionalAudio();
     media_expectations.ExpectNoVideo();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
   // Sanity check that the callee's description has a rejected video section.
   ASSERT_NE(nullptr, callee()->pc()->local_description());
   const ContentInfo* callee_video_content =
       GetFirstVideoContent(callee()->pc()->local_description()->description());
   ASSERT_NE(nullptr, callee_video_content);
   EXPECT_TRUE(callee_video_content->rejected);

   // Now negotiate with video and ensure negotiation succeeds, with video
   // frames and additional audio frames being received.
   callee()->AddVideoTrack();
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_video = 1;
     callee()->SetOfferAnswerOptions(options);
   } else {
     callee()->SetRemoteOfferHandler(nullptr);
     caller()->SetRemoteOfferHandler([this] {
       // The caller creates a new transceiver to receive video on when receiving
       // the offer, but by default it is send only.
       auto transceivers = caller()->pc()->GetTransceivers();
       ASSERT_EQ(3U, transceivers.size());
       ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
                 transceivers[2]->receiver()->media_type());
       transceivers[2]->sender()->SetTrack(caller()->CreateLocalVideoTrack());
       transceivers[2]->SetDirection(RtpTransceiverDirection::kSendRecv);
     });
   }
   callee()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   {
     // Expect additional audio frames to be received after the upgrade.
     MediaExpectations media_expectations;
     media_expectations.ExpectBidirectionalAudioAndVideo();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
 }

 // Simpler than the above test; just add an audio track to an established
 // video-only connection.
 TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Do initial offer/answer with just a video track.
   caller()->AddVideoTrack();
   callee()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Now add an audio track and do another offer/answer.
   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Ensure both audio and video frames are received end-to-end.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This test sets up a call that's transferred to a new caller with a different
 // DTLS fingerprint.
 TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Keep the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
   std::unique_ptr<PeerConnectionWrapper> original_peer(
       SetCallerPcWrapperAndReturnCurrent(
           CreatePeerConnectionWrapperWithAlternateKey().release()));
   // TODO(deadbeef): Why do we call Close here? That goes against the comment
   // directly above.
   original_peer->pc()->Close();

   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for some additional frames to be transmitted end-to-end.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This test sets up a call that's transferred to a new callee with a different
 // DTLS fingerprint.
 TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Keep the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
   std::unique_ptr<PeerConnectionWrapper> original_peer(
       SetCalleePcWrapperAndReturnCurrent(
           CreatePeerConnectionWrapperWithAlternateKey().release()));
   // TODO(deadbeef): Why do we call Close here? That goes against the comment
   // directly above.
   original_peer->pc()->Close();

   ConnectFakeSignaling();
   callee()->AddAudioVideoTracks();
   caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for some additional frames to be transmitted end-to-end.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This test sets up a non-bundled call and negotiates bundling at the same
 // time as starting an ICE restart. When bundling is in effect in the restart,
 // the DTLS-SRTP context should be successfully reset.
 TEST_P(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   // Remove the bundle group from the SDP received by the callee.
   callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
     desc->RemoveGroupByName("BUNDLE");
   });
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   {
     MediaExpectations media_expectations;
     media_expectations.ExpectBidirectionalAudioAndVideo();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
   // Now stop removing the BUNDLE group, and trigger an ICE restart.
   callee()->SetReceivedSdpMunger(nullptr);
   caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Expect additional frames to be received after the ICE restart.
   {
     MediaExpectations media_expectations;
     media_expectations.ExpectBidirectionalAudioAndVideo();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
 }

 // Test CVO (Coordination of Video Orientation). If a video source is rotated
 // and both peers support the CVO RTP header extension, the actual video frames
 // don't need to be encoded in different resolutions, since the rotation is
 // communicated through the RTP header extension.
 TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add rotated video tracks.
   caller()->AddTrack(
       caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
   callee()->AddTrack(
       callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));

   // Wait for video frames to be received by both sides.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                        callee()->min_video_frames_received_per_track() > 0,
                    kMaxWaitForFramesMs);

   // Ensure that the aspect ratio is unmodified.
   // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
   // not just assumed.
   EXPECT_EQ(4.0 / 3, caller()->local_rendered_aspect_ratio());
   EXPECT_EQ(4.0 / 3, caller()->rendered_aspect_ratio());
   EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio());
   EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio());
   // Ensure that the CVO bits were surfaced to the renderer.
   EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation());
   EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation());
 }

 // Test that when the CVO extension isn't supported, video is rotated the
 // old-fashioned way, by encoding rotated frames.
 TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add rotated video tracks.
   caller()->AddTrack(
       caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
   callee()->AddTrack(
       callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));

   // Remove the CVO extension from the offered SDP.
   callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
     cricket::VideoContentDescription* video =
         GetFirstVideoContentDescription(desc);
     video->ClearRtpHeaderExtensions();
   });
   // Wait for video frames to be received by both sides.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                        callee()->min_video_frames_received_per_track() > 0,
                    kMaxWaitForFramesMs);

   // Expect that the aspect ratio is inversed to account for the 90/270 degree
   // rotation.
   // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
   // not just assumed.
   EXPECT_EQ(3.0 / 4, caller()->local_rendered_aspect_ratio());
   EXPECT_EQ(3.0 / 4, caller()->rendered_aspect_ratio());
   EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio());
   EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio());
   // Expect that each endpoint is unaware of the rotation of the other endpoint.
   EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation());
   EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
 }

 // Test that if the answerer rejects the audio m= section, no audio is sent or
 // received, but video still can be.
 TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     // Only add video track for callee, and set offer_to_receive_audio to 0, so
     // it will reject the audio m= section completely.
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_audio = 0;
     callee()->SetOfferAnswerOptions(options);
   } else {
     // Stopping the audio RtpTransceiver will cause the media section to be
     // rejected in the answer.
     callee()->SetRemoteOfferHandler([this] {
       callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)->Stop();
     });
   }
   callee()->AddTrack(callee()->CreateLocalVideoTrack());
   // Do offer/answer and wait for successful end-to-end video frames.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalVideo();
   media_expectations.ExpectNoAudio();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   // Sanity check that the callee's description has a rejected audio section.
   ASSERT_NE(nullptr, callee()->pc()->local_description());
   const ContentInfo* callee_audio_content =
       GetFirstAudioContent(callee()->pc()->local_description()->description());
   ASSERT_NE(nullptr, callee_audio_content);
   EXPECT_TRUE(callee_audio_content->rejected);
   if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
     // The caller's transceiver should have stopped after receiving the answer.
     EXPECT_TRUE(caller()
                     ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
                     ->stopped());
   }
 }

 // Test that if the answerer rejects the video m= section, no video is sent or
 // received, but audio still can be.
 TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     // Only add audio track for callee, and set offer_to_receive_video to 0, so
     // it will reject the video m= section completely.
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_video = 0;
     callee()->SetOfferAnswerOptions(options);
   } else {
     // Stopping the video RtpTransceiver will cause the media section to be
     // rejected in the answer.
     callee()->SetRemoteOfferHandler([this] {
       callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
     });
   }
   callee()->AddTrack(callee()->CreateLocalAudioTrack());
   // Do offer/answer and wait for successful end-to-end audio frames.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudio();
   media_expectations.ExpectNoVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   // Sanity check that the callee's description has a rejected video section.
   ASSERT_NE(nullptr, callee()->pc()->local_description());
   const ContentInfo* callee_video_content =
       GetFirstVideoContent(callee()->pc()->local_description()->description());
   ASSERT_NE(nullptr, callee_video_content);
   EXPECT_TRUE(callee_video_content->rejected);
   if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
     // The caller's transceiver should have stopped after receiving the answer.
     EXPECT_TRUE(caller()
                     ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
                     ->stopped());
   }
 }

 // Test that if the answerer rejects both audio and video m= sections, nothing
 // bad happens.
 // TODO(deadbeef): Test that a data channel still works. Currently this doesn't
 // test anything but the fact that negotiation succeeds, which doesn't mean
 // much.
 TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
     // will reject both audio and video m= sections.
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_audio = 0;
     options.offer_to_receive_video = 0;
     callee()->SetOfferAnswerOptions(options);
   } else {
     callee()->SetRemoteOfferHandler([this] {
       // Stopping all transceivers will cause all media sections to be rejected.
       for (const auto& transceiver : callee()->pc()->GetTransceivers()) {
         transceiver->Stop();
       }
     });
   }
   // Do offer/answer and wait for stable signaling state.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Sanity check that the callee's description has rejected m= sections.
   ASSERT_NE(nullptr, callee()->pc()->local_description());
   const ContentInfo* callee_audio_content =
       GetFirstAudioContent(callee()->pc()->local_description()->description());
   ASSERT_NE(nullptr, callee_audio_content);
   EXPECT_TRUE(callee_audio_content->rejected);
   const ContentInfo* callee_video_content =
       GetFirstVideoContent(callee()->pc()->local_description()->description());
   ASSERT_NE(nullptr, callee_video_content);
   EXPECT_TRUE(callee_video_content->rejected);
 }

 // This test sets up an audio and video call between two parties. After the
 // call runs for a while, the caller sends an updated offer with video being
 // rejected. Once the re-negotiation is done, the video flow should stop and
 // the audio flow should continue.
 TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   {
     MediaExpectations media_expectations;
     media_expectations.ExpectBidirectionalAudioAndVideo();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
   // Renegotiate, rejecting the video m= section.
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     caller()->SetGeneratedSdpMunger(
         [](cricket::SessionDescription* description) {
           for (cricket::ContentInfo& content : description->contents()) {
             if (cricket::IsVideoContent(&content)) {
               content.rejected = true;
             }
           }
         });
   } else {
     caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
   }
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);

   // Sanity check that the caller's description has a rejected video section.
   ASSERT_NE(nullptr, caller()->pc()->local_description());
   const ContentInfo* caller_video_content =
       GetFirstVideoContent(caller()->pc()->local_description()->description());
   ASSERT_NE(nullptr, caller_video_content);
   EXPECT_TRUE(caller_video_content->rejected);
   // Wait for some additional audio frames to be received.
   {
     MediaExpectations media_expectations;
     media_expectations.ExpectBidirectionalAudio();
     media_expectations.ExpectNoVideo();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
 }

 // Do one offer/answer with audio, another that disables it (rejecting the m=
 // section), and another that re-enables it. Regression test for:
 // bugs.webrtc.org/6023
 TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Add audio track, do normal offer/answer.
   rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
       caller()->CreateLocalAudioTrack();
   rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
       caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Remove audio track, and set offer_to_receive_audio to false to cause the
   // m= section to be completely disabled, not just "recvonly".
   caller()->pc()->RemoveTrack(sender);
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio = 0;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Add the audio track again, expecting negotiation to succeed and frames to
   // flow.
   sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
   options.offer_to_receive_audio = 1;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudio();
   EXPECT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Basic end-to-end test, but without SSRC/MSID signaling. This functionality
 // is needed to support legacy endpoints.
 // TODO(deadbeef): When we support the MID extension and demuxing on MID, also
 // add a test for an end-to-end test without MID signaling either (basically,
 // the minimum acceptable SDP).
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add audio and video, testing that packets can be demuxed on payload type.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   // Remove SSRCs and MSIDs from the received offer SDP.
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Basic end-to-end test, without SSRC signaling. This means that the track
 // was created properly and frames are delivered when the MSIDs are communicated
 // with a=msid lines and no a=ssrc lines.
 TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
        EndToEndCallWithoutSsrcSignaling) {
   const char kStreamId[] = "streamId";
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add just audio tracks.
   caller()->AddTrack(caller()->CreateLocalAudioTrack(), {kStreamId});
   callee()->AddAudioTrack();

   // Remove SSRCs from the received offer SDP.
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndKeepMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudio();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Tests that video flows between multiple video tracks when SSRCs are not
 // signaled. This exercises the MID RTP header extension which is needed to
 // demux the incoming video tracks.
 TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
        EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddVideoTrack();
   caller()->AddVideoTrack();
   callee()->AddVideoTrack();
   callee()->AddVideoTrack();

   caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
   callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
   ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());

   // Expect video to be received in both directions on both tracks.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalVideo();
   EXPECT_TRUE(ExpectNewFrames(media_expectations));
 }

 TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLinePresent) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
   caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   auto callee_receivers = callee()->pc()->GetReceivers();
   ASSERT_EQ(2u, callee_receivers.size());
   EXPECT_TRUE(callee_receivers[0]->stream_ids().empty());
   EXPECT_TRUE(callee_receivers[1]->stream_ids().empty());
 }

 TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLineMissing) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
   caller()->AddVideoTrack();
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   auto callee_receivers = callee()->pc()->GetReceivers();
   ASSERT_EQ(2u, callee_receivers.size());
   ASSERT_EQ(1u, callee_receivers[0]->stream_ids().size());
   ASSERT_EQ(1u, callee_receivers[1]->stream_ids().size());
   EXPECT_EQ(callee_receivers[0]->stream_ids()[0],
             callee_receivers[1]->stream_ids()[0]);
   EXPECT_EQ(callee_receivers[0]->streams()[0],
             callee_receivers[1]->streams()[0]);
 }

 // Test that if two video tracks are sent (from caller to callee, in this test),
 // they're transmitted correctly end-to-end.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add one audio/video stream, and one video-only stream.
   caller()->AddAudioVideoTracks();
   caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_EQ(3u, callee()->pc()->GetReceivers().size());

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) {
   bool first = true;
   for (cricket::ContentInfo& content : desc->contents()) {
     if (first) {
       first = false;
       continue;
     }
     content.bundle_only = true;
   }
   first = true;
   for (cricket::TransportInfo& transport : desc->transport_infos()) {
     if (first) {
       first = false;
       continue;
     }
     transport.description.ice_ufrag.clear();
     transport.description.ice_pwd.clear();
     transport.description.connection_role = cricket::CONNECTIONROLE_NONE;
     transport.description.identity_fingerprint.reset(nullptr);
   }
 }

 // Test that if applying a true "max bundle" offer, which uses ports of 0,
 // "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and
 // "a=ice-pwd" for all but the audio "m=" section, negotiation still completes
 // successfully and media flows.
 // TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works.
 // TODO(deadbeef): Won't need this test once we start generating actual
 // standards-compliant SDP.
 TEST_P(PeerConnectionIntegrationTest,
        EndToEndCallWithSpecCompliantMaxBundleOffer) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   // Do the equivalent of setting the port to 0, adding a=bundle-only, and
   // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all
   // but the first m= section.
   callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test that we can receive the audio output level from a remote audio track.
 // TODO(deadbeef): Use a fake audio source and verify that the output level is
 // exactly what the source on the other side was configured with.
 TEST_P(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Just add an audio track.
   caller()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Get the audio output level stats. Note that the level is not available
   // until an RTCP packet has been received.
   EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0,
                    kMaxWaitForFramesMs);
 }

 // Test that an audio input level is reported.
 // TODO(deadbeef): Use a fake audio source and verify that the input level is
 // exactly what the source was configured with.
 TEST_P(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Just add an audio track.
   caller()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Get the audio input level stats. The level should be available very
   // soon after the test starts.
   EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0,
                    kMaxWaitForStatsMs);
 }

 // Test that we can get incoming byte counts from both audio and video tracks.
 TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   // Do offer/answer, wait for the callee to receive some frames.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   // Get a handle to the remote tracks created, so they can be used as GetStats
   // filters.
   for (const auto& receiver : callee()->pc()->GetReceivers()) {
     // We received frames, so we definitely should have nonzero "received bytes"
     // stats at this point.
     EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(),
               0);
   }
 }

 // Test that we can get outgoing byte counts from both audio and video tracks.
 TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   auto audio_track = caller()->CreateLocalAudioTrack();
   auto video_track = caller()->CreateLocalVideoTrack();
   caller()->AddTrack(audio_track);
   caller()->AddTrack(video_track);
   // Do offer/answer, wait for the callee to receive some frames.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   // The callee received frames, so we definitely should have nonzero "sent
   // bytes" stats at this point.
   EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0);
   EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
 }

 // Test that we can get capture start ntp time.
 TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();

   callee()->AddAudioTrack();

   // Do offer/answer, wait for the callee to receive some frames.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Get the remote audio track created on the receiver, so they can be used as
   // GetStats filters.
   auto receivers = callee()->pc()->GetReceivers();
   ASSERT_EQ(1u, receivers.size());
   auto remote_audio_track = receivers[0]->track();

   // Get the audio output level stats. Note that the level is not available
   // until an RTCP packet has been received.
   EXPECT_TRUE_WAIT(
       callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() >
           0,
       2 * kMaxWaitForFramesMs);
 }

 // Test that the track ID is associated with all local and remote SSRC stats
 // using the old GetStats() and more than 1 audio and more than 1 video track.
 // This is a regression test for crbug.com/906988
 TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
        OldGetStatsAssociatesTrackIdForManyMediaSections) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   auto audio_sender_1 = caller()->AddAudioTrack();
   auto video_sender_1 = caller()->AddVideoTrack();
   auto audio_sender_2 = caller()->AddAudioTrack();
   auto video_sender_2 = caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);

   std::vector<std::string> track_ids = {
       audio_sender_1->track()->id(), video_sender_1->track()->id(),
       audio_sender_2->track()->id(), video_sender_2->track()->id()};

   auto caller_stats = caller()->OldGetStats();
   EXPECT_THAT(caller_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
   auto callee_stats = callee()->OldGetStats();
   EXPECT_THAT(callee_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
 }

 // Test that the new GetStats() returns stats for all outgoing/incoming streams
 // with the correct track IDs if there are more than one audio and more than one
 // video senders/receivers.
 TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   auto audio_sender_1 = caller()->AddAudioTrack();
   auto video_sender_1 = caller()->AddVideoTrack();
   auto audio_sender_2 = caller()->AddAudioTrack();
   auto video_sender_2 = caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);

   std::vector<std::string> track_ids = {
       audio_sender_1->track()->id(), video_sender_1->track()->id(),
       audio_sender_2->track()->id(), video_sender_2->track()->id()};

   rtc::scoped_refptr<const webrtc::RTCStatsReport> caller_report =
       caller()->NewGetStats();
   ASSERT_TRUE(caller_report);
   auto outbound_stream_stats =
       caller_report->GetStatsOfType<webrtc::RTCOutboundRTPStreamStats>();
   ASSERT_EQ(4u, outbound_stream_stats.size());
   std::vector<std::string> outbound_track_ids;
   for (const auto& stat : outbound_stream_stats) {
     ASSERT_TRUE(stat->bytes_sent.is_defined());
     EXPECT_LT(0u, *stat->bytes_sent);
     ASSERT_TRUE(stat->track_id.is_defined());
     const auto* track_stat =
         caller_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
     ASSERT_TRUE(track_stat);
     outbound_track_ids.push_back(*track_stat->track_identifier);
   }
   EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));

   rtc::scoped_refptr<const webrtc::RTCStatsReport> callee_report =
       callee()->NewGetStats();
   ASSERT_TRUE(callee_report);
   auto inbound_stream_stats =
       callee_report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
   ASSERT_EQ(4u, inbound_stream_stats.size());
   std::vector<std::string> inbound_track_ids;
   for (const auto& stat : inbound_stream_stats) {
     ASSERT_TRUE(stat->bytes_received.is_defined());
     EXPECT_LT(0u, *stat->bytes_received);
     ASSERT_TRUE(stat->track_id.is_defined());
     const auto* track_stat =
         callee_report->GetAs<webrtc::RTCMediaStreamTrackStats>(*stat->track_id);
     ASSERT_TRUE(track_stat);
     inbound_track_ids.push_back(*track_stat->track_identifier);
   }
   EXPECT_THAT(inbound_track_ids, UnorderedElementsAreArray(track_ids));
 }

 // Test that we can get stats (using the new stats implementation) for
 // unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
 // SDP.
 TEST_P(PeerConnectionIntegrationTest,
        GetStatsForUnsignaledStreamWithNewStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
   // Remove SSRCs and MSIDs from the received offer SDP.
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudio(1);
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   // We received a frame, so we should have nonzero "bytes received" stats for
   // the unsignaled stream, if stats are working for it.
   rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
       callee()->NewGetStats();
   ASSERT_NE(nullptr, report);
   auto inbound_stream_stats =
       report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
   ASSERT_EQ(1U, inbound_stream_stats.size());
   ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
   ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
   ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
 }

 // Same as above but for the legacy stats implementation.
 TEST_P(PeerConnectionIntegrationTest,
        GetStatsForUnsignaledStreamWithOldStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
   // Remove SSRCs and MSIDs from the received offer SDP.
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Note that, since the old stats implementation associates SSRCs with tracks
   // using SDP, when SSRCs aren't signaled in SDP these stats won't have an
   // associated track ID. So we can't use the track "selector" argument.
   //
   // Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to
   // return cached stats if not enough time has passed since the last update.
   EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0,
                    kDefaultTimeout);
 }

 // Test that we can successfully get the media related stats (audio level
 // etc.) for the unsignaled stream.
 TEST_P(PeerConnectionIntegrationTest,
        GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   // Remove SSRCs and MSIDs from the received offer SDP.
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudio(1);
   media_expectations.CalleeExpectsSomeVideo(1);
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
       callee()->NewGetStats();
   ASSERT_NE(nullptr, report);

   auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
   auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats);
   ASSERT_GE(audio_index, 0);
   EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined());
 }

 // Helper for test below.
 void ModifySsrcs(cricket::SessionDescription* desc) {
   for (ContentInfo& content : desc->contents()) {
     for (StreamParams& stream :
          content.media_description()->mutable_streams()) {
       for (uint32_t& ssrc : stream.ssrcs) {
         ssrc = rtc::CreateRandomId();
       }
     }
   }
 }

 // Test that the "RTCMediaSteamTrackStats"  object is updated correctly when
 // SSRCs are unsignaled, and the SSRC of the received (audio) stream changes.
 // This should result in two "RTCInboundRTPStreamStats", but only one
 // "RTCMediaStreamTrackStats", whose counters go up continuously rather than
 // being reset to 0 once the SSRC change occurs.
 //
 // Regression test for this bug:
 // https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
 //
 // The bug causes the track stats to only represent one of the two streams:
 // whichever one has the higher SSRC. So with this bug, there was a 50% chance
 // that the track stat counters would reset to 0 when the new stream is
 // received, and a 50% chance that they'll stop updating (while
 // "concealed_samples" continues increasing, due to silence being generated for
 // the inactive stream).
 TEST_P(PeerConnectionIntegrationTest,
        TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
   // Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint
   // that doesn't signal SSRCs (from the callee's perspective).
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for 50 audio frames (500ms of audio) to be received by the callee.
   {
     MediaExpectations media_expectations;
     media_expectations.CalleeExpectsSomeAudio(50);
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
   // Some audio frames were received, so we should have nonzero "samples
   // received" for the track.
   rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
       callee()->NewGetStats();
   ASSERT_NE(nullptr, report);
   auto track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
   ASSERT_EQ(1U, track_stats.size());
   ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
   ASSERT_GT(*track_stats[0]->total_samples_received, 0U);
   // uint64_t prev_samples_received = *track_stats[0]->total_samples_received;

   // Create a new offer and munge it to cause the caller to use a new SSRC.
   caller()->SetGeneratedSdpMunger(ModifySsrcs);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for 25 more audio frames (250ms of audio) to be received, from the new
   // SSRC.
   {
     MediaExpectations media_expectations;
     media_expectations.CalleeExpectsSomeAudio(25);
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }

   report = callee()->NewGetStats();
   ASSERT_NE(nullptr, report);
   track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
   ASSERT_EQ(1U, track_stats.size());
   ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
   // The "total samples received" stat should only be greater than it was
   // before.
   // TODO(deadbeef): Uncomment this assertion once the bug is completely fixed.
   // Right now, the new SSRC will cause the counters to reset to 0.
   // EXPECT_GT(*track_stats[0]->total_samples_received, prev_samples_received);

   // Additionally, the percentage of concealed samples (samples generated to
   // conceal packet loss) should be less than 50%. If it's greater, that's a
   // good sign that we're seeing stats from the old stream that's no longer
   // receiving packets, and is generating concealed samples of silence.
   constexpr double kAcceptableConcealedSamplesPercentage = 0.50;
   ASSERT_TRUE(track_stats[0]->concealed_samples.is_defined());
   EXPECT_LT(*track_stats[0]->concealed_samples,
             *track_stats[0]->total_samples_received *
                 kAcceptableConcealedSamplesPercentage);

   // Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a
   // sanity check that the SSRC really changed.
   // TODO(deadbeef): This isn't working right now, because we're not returning
   // *any* stats for the inactive stream. Uncomment when the bug is completely
   // fixed.
   // auto inbound_stream_stats =
   //     report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
   // ASSERT_EQ(2U, inbound_stream_stats.size());
 }

 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
   PeerConnectionFactory::Options dtls_10_options;
   dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
                                                       dtls_10_options));
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for some frames to be received in each
   // direction.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated.
 TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
   PeerConnectionFactory::Options dtls_10_options;
   dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
                                                       dtls_10_options));
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
                        caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
                    kDefaultTimeout);
   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                  caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
   // TODO(bugs.webrtc.org/9456): Fix it.
   EXPECT_EQ(1, webrtc::metrics::NumEvents(
                    "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                    kDefaultSrtpCryptoSuite));
 }

 // Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
 TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
   PeerConnectionFactory::Options dtls_12_options;
   dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
                                                       dtls_12_options));
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
                        caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
                    kDefaultTimeout);
   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                  caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
   // TODO(bugs.webrtc.org/9456): Fix it.
   EXPECT_EQ(1, webrtc::metrics::NumEvents(
                    "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                    kDefaultSrtpCryptoSuite));
 }

 // Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
 // callee only supports 1.0.
 TEST_P(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
   PeerConnectionFactory::Options caller_options;
   caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   PeerConnectionFactory::Options callee_options;
   callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for some frames to be received in each
   // direction.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the
 // callee supports 1.2.
 TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
   PeerConnectionFactory::Options caller_options;
   caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
   PeerConnectionFactory::Options callee_options;
   callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for some frames to be received in each
   // direction.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // The three tests below verify that "enable_aes128_sha1_32_crypto_cipher"
 // works as expected; the cipher should only be used if enabled by both sides.
 TEST_P(PeerConnectionIntegrationTest,
        Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
   PeerConnectionFactory::Options caller_options;
   caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
   PeerConnectionFactory::Options callee_options;
   callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
       false;
   int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
   TestNegotiatedCipherSuite(caller_options, callee_options,
                             expected_cipher_suite);
 }

 TEST_P(PeerConnectionIntegrationTest,
        Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
   PeerConnectionFactory::Options caller_options;
   caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
       false;
   PeerConnectionFactory::Options callee_options;
   callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
   int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
   TestNegotiatedCipherSuite(caller_options, callee_options,
                             expected_cipher_suite);
 }

 TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
   PeerConnectionFactory::Options caller_options;
   caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
   PeerConnectionFactory::Options callee_options;
   callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
   int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
   TestNegotiatedCipherSuite(caller_options, callee_options,
                             expected_cipher_suite);
 }

 // Test that a non-GCM cipher is used if both sides only support non-GCM.
 TEST_P(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
   bool local_gcm_enabled = false;
   bool remote_gcm_enabled = false;
   int expected_cipher_suite = kDefaultSrtpCryptoSuite;
   TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                     expected_cipher_suite);
 }

 // Test that a GCM cipher is used if both ends support it.
 TEST_P(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) {
   bool local_gcm_enabled = true;
   bool remote_gcm_enabled = true;
   int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm;
   TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                     expected_cipher_suite);
 }

 // Test that GCM isn't used if only the offerer supports it.
 TEST_P(PeerConnectionIntegrationTest,
        NonGcmCipherUsedWhenOnlyCallerSupportsGcm) {
   bool local_gcm_enabled = true;
   bool remote_gcm_enabled = false;
   int expected_cipher_suite = kDefaultSrtpCryptoSuite;
   TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                     expected_cipher_suite);
 }

 // Test that GCM isn't used if only the answerer supports it.
 TEST_P(PeerConnectionIntegrationTest,
        NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) {
   bool local_gcm_enabled = false;
   bool remote_gcm_enabled = true;
   int expected_cipher_suite = kDefaultSrtpCryptoSuite;
   TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                     expected_cipher_suite);
 }

 // Verify that media can be transmitted end-to-end when GCM crypto suites are
 // enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported,
 // only verify that a GCM cipher is negotiated, and not necessarily that SRTP
 // works with it.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
   PeerConnectionFactory::Options gcm_options;
   gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true;
   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for some frames to be received in each
   // direction.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This test sets up a call between two parties with audio, video and an RTP
 // data channel.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.enable_rtp_data_channel = true;
   rtc_config.enable_dtls_srtp = false;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
   ConnectFakeSignaling();
   // Expect that data channel created on caller side will show up for callee as
   // well.
   caller()->CreateDataChannel();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Ensure the existence of the RTP data channel didn't impede audio/video.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_NE(nullptr, callee()->data_channel());
   EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Ensure data can be sent in both directions.
   std::string data = "hello world";
   SendRtpDataWithRetries(caller()->data_channel(), data, 5);
   EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                  kDefaultTimeout);
   SendRtpDataWithRetries(callee()->data_channel(), data, 5);
   EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                  kDefaultTimeout);
 }

 // Ensure that an RTP data channel is signaled as closed for the caller when
 // the callee rejects it in a subsequent offer.
 TEST_P(PeerConnectionIntegrationTest,
        RtpDataChannelSignaledClosedInCalleeOffer) {
   // Same procedure as above test.
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.enable_rtp_data_channel = true;
   rtc_config.enable_dtls_srtp = false;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
   ConnectFakeSignaling();
   caller()->CreateDataChannel();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_NE(nullptr, callee()->data_channel());
   ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Close the data channel on the callee, and do an updated offer/answer.
   callee()->data_channel()->Close();
   callee()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   EXPECT_FALSE(caller()->data_observer()->IsOpen());
   EXPECT_FALSE(callee()->data_observer()->IsOpen());
 }

 // Tests that data is buffered in an RTP data channel until an observer is
 // registered for it.
 //
 // NOTE: RTP data channels can receive data before the underlying
 // transport has detected that a channel is writable and thus data can be
 // received before the data channel state changes to open. That is hard to test
 // but the same buffering is expected to be used in that case.
 TEST_P(PeerConnectionIntegrationTest,
        DataBufferedUntilRtpDataChannelObserverRegistered) {
   // Use fake clock and simulated network delay so that we predictably can wait
   // until an SCTP message has been delivered without "sleep()"ing.
   rtc::ScopedFakeClock fake_clock;
   // Some things use a time of "0" as a special value, so we need to start out
   // the fake clock at a nonzero time.
   // TODO(deadbeef): Fix this.
   fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
   virtual_socket_server()->set_delay_mean(5);  // 5 ms per hop.
   virtual_socket_server()->UpdateDelayDistribution();

   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.enable_rtp_data_channel = true;
   rtc_config.enable_dtls_srtp = false;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
   ConnectFakeSignaling();
   caller()->CreateDataChannel();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE(caller()->data_channel() != nullptr);
   ASSERT_TRUE_SIMULATED_WAIT(callee()->data_channel() != nullptr,
                              kDefaultTimeout, fake_clock);
   ASSERT_TRUE_SIMULATED_WAIT(caller()->data_observer()->IsOpen(),
                              kDefaultTimeout, fake_clock);
   ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen,
                            callee()->data_channel()->state(), kDefaultTimeout,
                            fake_clock);

   // Unregister the observer which is normally automatically registered.
   callee()->data_channel()->UnregisterObserver();
   // Send data and advance fake clock until it should have been received.
   std::string data = "hello world";
   caller()->data_channel()->Send(DataBuffer(data));
   SIMULATED_WAIT(false, 50, fake_clock);

   // Attach data channel and expect data to be received immediately. Note that
   // EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any
   // further, but data can be received even if the callback is asynchronous.
   MockDataChannelObserver new_observer(callee()->data_channel());
   EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout,
                            fake_clock);
   // Closing the PeerConnections destroys the ports before the ScopedFakeClock.
   // If this is not done a DCHECK can be hit in ports.cc, because a large
   // negative number is calculated for the rtt due to the global clock changing.
   ClosePeerConnections();
 }

 // This test sets up a call between two parties with audio, video and but only
 // the caller client supports RTP data channels.
 TEST_P(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
   PeerConnectionInterface::RTCConfiguration rtc_config_1;
   rtc_config_1.enable_rtp_data_channel = true;
   // Must disable DTLS to make negotiation succeed.
   rtc_config_1.enable_dtls_srtp = false;
   PeerConnectionInterface::RTCConfiguration rtc_config_2;
   rtc_config_2.enable_dtls_srtp = false;
   rtc_config_2.enable_dtls_srtp = false;
   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(rtc_config_1, rtc_config_2));
   ConnectFakeSignaling();
   caller()->CreateDataChannel();
   ASSERT_TRUE(caller()->data_channel() != nullptr);
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // The caller should still have a data channel, but it should be closed, and
   // one should ever have been created for the callee.
   EXPECT_TRUE(caller()->data_channel() != nullptr);
   EXPECT_FALSE(caller()->data_observer()->IsOpen());
   EXPECT_EQ(nullptr, callee()->data_channel());
 }

 // This test sets up a call between two parties with audio, and video. When
 // audio and video is setup and flowing, an RTP data channel is negotiated.
 TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.enable_rtp_data_channel = true;
   rtc_config.enable_dtls_srtp = false;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
   ConnectFakeSignaling();
   // Do initial offer/answer with audio/video.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Create data channel and do new offer and answer.
   caller()->CreateDataChannel();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_NE(nullptr, callee()->data_channel());
   EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
   // Ensure data can be sent in both directions.
   std::string data = "hello world";
   SendRtpDataWithRetries(caller()->data_channel(), data, 5);
   EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                  kDefaultTimeout);
   SendRtpDataWithRetries(callee()->data_channel(), data, 5);
   EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                  kDefaultTimeout);
 }

 #ifdef HAVE_SCTP

 // This test sets up a call between two parties with audio, video and an SCTP
 // data channel.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Expect that data channel created on caller side will show up for callee as
   // well.
   caller()->CreateDataChannel();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Ensure the existence of the SCTP data channel didn't impede audio/video.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
   // Caller data channel should already exist (it created one). Callee data
   // channel may not exist yet, since negotiation happens in-band, not in SDP.
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Ensure data can be sent in both directions.
   std::string data = "hello world";
   caller()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                  kDefaultTimeout);
   callee()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                  kDefaultTimeout);
 }

 // Ensure that when the callee closes an SCTP data channel, the closing
 // procedure results in the data channel being closed for the caller as well.
 TEST_P(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
   // Same procedure as above test.
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->CreateDataChannel();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Close the data channel on the callee side, and wait for it to reach the
   // "closed" state on both sides.
   callee()->data_channel()->Close();
   EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
 }

 TEST_P(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   webrtc::DataChannelInit init;
   init.id = 53;
   init.maxRetransmits = 52;
   caller()->CreateDataChannel("data-channel", &init);
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
   // Since "negotiated" is false, the "id" parameter should be ignored.
   EXPECT_NE(init.id, callee()->data_channel()->id());
   EXPECT_EQ("data-channel", callee()->data_channel()->label());
   EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
   EXPECT_FALSE(callee()->data_channel()->negotiated());
 }

 // Test usrsctp's ability to process unordered data stream, where data actually
 // arrives out of order using simulated delays. Previously there have been some
 // bugs in this area.
 TEST_P(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
   // Introduce random network delays.
   // Otherwise it's not a true "unordered" test.
   virtual_socket_server()->set_delay_mean(20);
   virtual_socket_server()->set_delay_stddev(5);
   virtual_socket_server()->UpdateDelayDistribution();
   // Normal procedure, but with unordered data channel config.
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   webrtc::DataChannelInit init;
   init.ordered = false;
   caller()->CreateDataChannel(&init);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   static constexpr int kNumMessages = 100;
   // Deliberately chosen to be larger than the MTU so messages get fragmented.
   static constexpr size_t kMaxMessageSize = 4096;
   // Create and send random messages.
   std::vector<std::string> sent_messages;
   for (int i = 0; i < kNumMessages; ++i) {
     size_t length =
         (rand() % kMaxMessageSize) + 1;  // NOLINT (rand_r instead of rand)
     std::string message;
     ASSERT_TRUE(rtc::CreateRandomString(length, &message));
     caller()->data_channel()->Send(DataBuffer(message));
     callee()->data_channel()->Send(DataBuffer(message));
     sent_messages.push_back(message);
   }

   // Wait for all messages to be received.
   EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
                  caller()->data_observer()->received_message_count(),
                  kDefaultTimeout);
   EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
                  callee()->data_observer()->received_message_count(),
                  kDefaultTimeout);

   // Sort and compare to make sure none of the messages were corrupted.
   std::vector<std::string> caller_received_messages =
       caller()->data_observer()->messages();
   std::vector<std::string> callee_received_messages =
       callee()->data_observer()->messages();
   absl::c_sort(sent_messages);
   absl::c_sort(caller_received_messages);
   absl::c_sort(callee_received_messages);
   EXPECT_EQ(sent_messages, caller_received_messages);
   EXPECT_EQ(sent_messages, callee_received_messages);
 }

 // This test sets up a call between two parties with audio, and video. When
 // audio and video are setup and flowing, an SCTP data channel is negotiated.
 TEST_P(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Do initial offer/answer with audio/video.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Create data channel and do new offer and answer.
   caller()->CreateDataChannel();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Caller data channel should already exist (it created one). Callee data
   // channel may not exist yet, since negotiation happens in-band, not in SDP.
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
   // Ensure data can be sent in both directions.
   std::string data = "hello world";
   caller()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                  kDefaultTimeout);
   callee()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                  kDefaultTimeout);
 }

 // Set up a connection initially just using SCTP data channels, later upgrading
 // to audio/video, ensuring frames are received end-to-end. Effectively the
 // inverse of the test above.
 // This was broken in M57; see https://crbug.com/711243
 TEST_P(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Do initial offer/answer with just data channel.
   caller()->CreateDataChannel();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait until data can be sent over the data channel.
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Do subsequent offer/answer with two-way audio and video. Audio and video
   // should end up bundled on the DTLS/ICE transport already used for data.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
   cricket::SctpDataContentDescription* dcd_offer =
       GetFirstSctpDataContentDescription(desc);
   ASSERT_TRUE(dcd_offer);
   dcd_offer->set_use_sctpmap(false);
   dcd_offer->set_protocol("UDP/DTLS/SCTP");
 }

 // Test that the data channel works when a spec-compliant SCTP m= section is
 // offered (using "a=sctp-port" instead of "a=sctpmap", and using
 // "UDP/DTLS/SCTP" as the protocol).
 TEST_P(PeerConnectionIntegrationTest,
        DataChannelWorksWhenSpecCompliantSctpOfferReceived) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->CreateDataChannel();
   caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Ensure data can be sent in both directions.
   std::string data = "hello world";
   caller()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                  kDefaultTimeout);
   callee()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                  kDefaultTimeout);
 }

 #endif  // HAVE_SCTP

 // This test sets up a call between two parties with a media transport data
 // channel.
 TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelEndToEnd) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
   rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
   rtc_config.use_media_transport_for_data_channels = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   // Expect that data channel created on caller side will show up for callee as
   // well.
   caller()->CreateDataChannel();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that the media transport is ready.
   loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
   loopback_media_transports()->FlushAsyncInvokes();

   // Caller data channel should already exist (it created one). Callee data
   // channel may not exist yet, since negotiation happens in-band, not in SDP.
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Ensure data can be sent in both directions.
   std::string data = "hello world";
   caller()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                  kDefaultTimeout);
   callee()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                  kDefaultTimeout);
 }

 // Ensure that when the callee closes a media transport data channel, the
 // closing procedure results in the data channel being closed for the caller
 // as well.
 TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelCalleeCloses) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.use_media_transport_for_data_channels = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   // Create a data channel on the caller and signal it to the callee.
   caller()->CreateDataChannel();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that the media transport is ready.
   loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
   loopback_media_transports()->FlushAsyncInvokes();

   // Data channels exist and open on both ends of the connection.
   ASSERT_NE(nullptr, caller()->data_channel());
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);

   // Close the data channel on the callee side, and wait for it to reach the
   // "closed" state on both sides.
   callee()->data_channel()->Close();
   EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
   EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
 }

 TEST_P(PeerConnectionIntegrationTest,
        MediaTransportDataChannelConfigSentToOtherSide) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.use_media_transport_for_data_channels = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   // Create a data channel with a non-default configuration and signal it to the
   // callee.
   webrtc::DataChannelInit init;
   init.id = 53;
   init.maxRetransmits = 52;
   caller()->CreateDataChannel("data-channel", &init);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that the media transport is ready.
   loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
   loopback_media_transports()->FlushAsyncInvokes();

   // Ensure that the data channel exists on the callee with the correct
   // configuration.
   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
   // Since "negotiate" is false, the "id" parameter is ignored.
   EXPECT_NE(init.id, callee()->data_channel()->id());
   EXPECT_EQ("data-channel", callee()->data_channel()->label());
   EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
   EXPECT_FALSE(callee()->data_channel()->negotiated());
 }

 TEST_P(PeerConnectionIntegrationTest, MediaTransportOfferUpgrade) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
   rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
   rtc_config.use_media_transport = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   // Do initial offer/answer with just a video track.
   caller()->AddVideoTrack();
   callee()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that the media transport is ready.
   loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
   loopback_media_transports()->FlushAsyncInvokes();

   // Now add an audio track and do another offer/answer.
   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure both audio and video frames are received end-to-end.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   // The second offer should not have generated another media transport.
   // Media transport was kept alive, and was not recreated.
   EXPECT_EQ(1, loopback_media_transports()->first_factory_transport_count());
   EXPECT_EQ(1, loopback_media_transports()->second_factory_transport_count());
 }

 TEST_P(PeerConnectionIntegrationTest, MediaTransportOfferUpgradeOnTheCallee) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
   rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
   rtc_config.use_media_transport = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   // Do initial offer/answer with just a video track.
   caller()->AddVideoTrack();
   callee()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that the media transport is ready.
   loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
   loopback_media_transports()->FlushAsyncInvokes();

   // Now add an audio track and do another offer/answer.
   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   callee()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure both audio and video frames are received end-to-end.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   // The second offer should not have generated another media transport.
   // Media transport was kept alive, and was not recreated.
   EXPECT_EQ(1, loopback_media_transports()->first_factory_transport_count());
   EXPECT_EQ(1, loopback_media_transports()->second_factory_transport_count());
 }

 TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalAudio) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
   rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
   rtc_config.use_media_transport = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   // Start offer/answer exchange and wait for it to complete.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that the media transport is ready.
   loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
   loopback_media_transports()->FlushAsyncInvokes();

   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudio();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   webrtc::MediaTransportPair::Stats first_stats =
       loopback_media_transports()->FirstStats();
   webrtc::MediaTransportPair::Stats second_stats =
       loopback_media_transports()->SecondStats();

   EXPECT_GT(first_stats.received_audio_frames, 0);
   EXPECT_GE(second_stats.sent_audio_frames, first_stats.received_audio_frames);

   EXPECT_GT(second_stats.received_audio_frames, 0);
   EXPECT_GE(first_stats.sent_audio_frames, second_stats.received_audio_frames);
 }

 TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalVideo) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.use_media_transport = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   caller()->AddVideoTrack();
   callee()->AddVideoTrack();
   // Start offer/answer exchange and wait for it to complete.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that the media transport is ready.
   loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
   loopback_media_transports()->FlushAsyncInvokes();

   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));

   webrtc::MediaTransportPair::Stats first_stats =
       loopback_media_transports()->FirstStats();
   webrtc::MediaTransportPair::Stats second_stats =
       loopback_media_transports()->SecondStats();

   EXPECT_GT(first_stats.received_video_frames, 0);
   EXPECT_GE(second_stats.sent_video_frames, first_stats.received_video_frames);

   EXPECT_GT(second_stats.received_video_frames, 0);
   EXPECT_GE(first_stats.sent_video_frames, second_stats.received_video_frames);
 }

 TEST_P(PeerConnectionIntegrationTest,
        MediaTransportDataChannelUsesRtpBidirectionalVideo) {
   PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.use_media_transport = false;
   rtc_config.use_media_transport_for_data_channels = true;
   rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
       rtc_config, rtc_config, loopback_media_transports()->first_factory(),
       loopback_media_transports()->second_factory()));
   ConnectFakeSignaling();

   caller()->AddVideoTrack();
   callee()->AddVideoTrack();
   // Start offer/answer exchange and wait for it to complete.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test that the ICE connection and gathering states eventually reach
 // "complete".
 TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Do normal offer/answer.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
                  caller()->ice_gathering_state(), kMaxWaitForFramesMs);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
                  callee()->ice_gathering_state(), kMaxWaitForFramesMs);
   // After the best candidate pair is selected and all candidates are signaled,
   // the ICE connection state should reach "complete".
   // TODO(deadbeef): Currently, the ICE "controlled" agent (the
   // answerer/"callee" by default) only reaches "connected". When this is
   // fixed, this test should be updated.
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);
 }

 constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
                                 cricket::PORTALLOCATOR_DISABLE_RELAY |
                                 cricket::PORTALLOCATOR_DISABLE_TCP;

 // Use a mock resolver to resolve the hostname back to the original IP on both
 // sides and check that the ICE connection connects.
 TEST_P(PeerConnectionIntegrationTest,
        IceStatesReachCompletionWithRemoteHostname) {
   auto caller_resolver_factory =
       absl::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
   auto callee_resolver_factory =
       absl::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
   NiceMock<rtc::MockAsyncResolver> callee_async_resolver;
   NiceMock<rtc::MockAsyncResolver> caller_async_resolver;

   // This also verifies that the injected AsyncResolverFactory is used by
   // P2PTransportChannel.
   EXPECT_CALL(*caller_resolver_factory, Create())
       .WillOnce(Return(&caller_async_resolver));
   webrtc::PeerConnectionDependencies caller_deps(nullptr);
   caller_deps.async_resolver_factory = std::move(caller_resolver_factory);

   EXPECT_CALL(*callee_resolver_factory, Create())
       .WillOnce(Return(&callee_async_resolver));
   webrtc::PeerConnectionDependencies callee_deps(nullptr);
   callee_deps.async_resolver_factory = std::move(callee_resolver_factory);

   PeerConnectionInterface::RTCConfiguration config;
   config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
   config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;

   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
       config, std::move(caller_deps), config, std::move(callee_deps)));

   caller()->SetRemoteAsyncResolver(&callee_async_resolver);
   callee()->SetRemoteAsyncResolver(&caller_async_resolver);

   // Enable hostname candidates with mDNS names.
   caller()->SetMdnsResponder(
       absl::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
   callee()->SetMdnsResponder(
       absl::make_unique<webrtc::FakeMdnsResponder>(network_thread()));

   SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts);

   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);

   EXPECT_EQ(1, webrtc::metrics::NumEvents(
                    "WebRTC.PeerConnection.CandidatePairType_UDP",
                    webrtc::kIceCandidatePairHostNameHostName));
 }

 // Test that firewalling the ICE connection causes the clients to identify the
 // disconnected state and then removing the firewall causes them to reconnect.
 class PeerConnectionIntegrationIceStatesTest
     : public PeerConnectionIntegrationBaseTest,
       public ::testing::WithParamInterface<
           std::tuple<SdpSemantics, std::tuple<std::string, uint32_t>>> {
  protected:
   PeerConnectionIntegrationIceStatesTest()
       : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())) {
     port_allocator_flags_ = std::get<1>(std::get<1>(GetParam()));
   }

   void StartStunServer(const SocketAddress& server_address) {
     stun_server_.reset(
         cricket::TestStunServer::Create(network_thread(), server_address));
   }

   bool TestIPv6() {
     return (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6);
   }

   void SetPortAllocatorFlags() {
     PeerConnectionIntegrationBaseTest::SetPortAllocatorFlags(
         port_allocator_flags_, port_allocator_flags_);
   }

   std::vector<SocketAddress> CallerAddresses() {
     std::vector<SocketAddress> addresses;
     addresses.push_back(SocketAddress("1.1.1.1", 0));
     if (TestIPv6()) {
       addresses.push_back(SocketAddress("1111:0:a:b:c:d:e:f", 0));
     }
     return addresses;
   }

   std::vector<SocketAddress> CalleeAddresses() {
     std::vector<SocketAddress> addresses;
     addresses.push_back(SocketAddress("2.2.2.2", 0));
     if (TestIPv6()) {
       addresses.push_back(SocketAddress("2222:0:a:b:c:d:e:f", 0));
     }
     return addresses;
   }

   void SetUpNetworkInterfaces() {
     // Remove the default interfaces added by the test infrastructure.
     caller()->network_manager()->RemoveInterface(kDefaultLocalAddress);
     callee()->network_manager()->RemoveInterface(kDefaultLocalAddress);

     // Add network addresses for test.
     for (const auto& caller_address : CallerAddresses()) {
       caller()->network_manager()->AddInterface(caller_address);
     }
     for (const auto& callee_address : CalleeAddresses()) {
       callee()->network_manager()->AddInterface(callee_address);
     }
   }

  private:
   uint32_t port_allocator_flags_;
   std::unique_ptr<cricket::TestStunServer> stun_server_;
 };

 // Tests that the PeerConnection goes through all the ICE gathering/connection
 // states over the duration of the call. This includes Disconnected and Failed
 // states, induced by putting a firewall between the peers and waiting for them
 // to time out.
 TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) {
   rtc::ScopedFakeClock fake_clock;
   // Some things use a time of "0" as a special value, so we need to start out
   // the fake clock at a nonzero time.
   fake_clock.AdvanceTime(TimeDelta::seconds(1));

   const SocketAddress kStunServerAddress =
       SocketAddress("99.99.99.1", cricket::STUN_SERVER_PORT);
   StartStunServer(kStunServerAddress);

   PeerConnectionInterface::RTCConfiguration config;
   PeerConnectionInterface::IceServer ice_stun_server;
   ice_stun_server.urls.push_back(
       "stun:" + kStunServerAddress.HostAsURIString() + ":" +
       kStunServerAddress.PortAsString());
   config.servers.push_back(ice_stun_server);

   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
   ConnectFakeSignaling();
   SetPortAllocatorFlags();
   SetUpNetworkInterfaces();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();

   // Initial state before anything happens.
   ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew,
             caller()->ice_gathering_state());
   ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
             caller()->ice_connection_state());
   ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
             caller()->standardized_ice_connection_state());

   // Start the call by creating the offer, setting it as the local description,
   // then sending it to the peer who will respond with an answer. This happens
   // asynchronously so that we can watch the states as it runs in the
   // background.
   caller()->CreateAndSetAndSignalOffer();

   ASSERT_EQ(PeerConnectionInterface::kIceConnectionCompleted,
             caller()->ice_connection_state());
   ASSERT_EQ(PeerConnectionInterface::kIceConnectionCompleted,
             caller()->standardized_ice_connection_state());

   // Verify that the observer was notified of the intermediate transitions.
   EXPECT_THAT(caller()->ice_connection_state_history(),
               ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
                           PeerConnectionInterface::kIceConnectionConnected,
                           PeerConnectionInterface::kIceConnectionCompleted));
   EXPECT_THAT(caller()->standardized_ice_connection_state_history(),
               ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
                           PeerConnectionInterface::kIceConnectionConnected,
                           PeerConnectionInterface::kIceConnectionCompleted));
   EXPECT_THAT(
       caller()->peer_connection_state_history(),
       ElementsAre(PeerConnectionInterface::PeerConnectionState::kConnecting,
                   PeerConnectionInterface::PeerConnectionState::kConnected));
   EXPECT_THAT(caller()->ice_gathering_state_history(),
               ElementsAre(PeerConnectionInterface::kIceGatheringGathering,
                           PeerConnectionInterface::kIceGatheringComplete));

   // Block connections to/from the caller and wait for ICE to become
   // disconnected.
   for (const auto& caller_address : CallerAddresses()) {
     firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
   }
   RTC_LOG(LS_INFO) << "Firewall rules applied";
   ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
                            caller()->ice_connection_state(), kDefaultTimeout,
                            fake_clock);
   ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
                            caller()->standardized_ice_connection_state(),
                            kDefaultTimeout, fake_clock);

   // Let ICE re-establish by removing the firewall rules.
   firewall()->ClearRules();
   RTC_LOG(LS_INFO) << "Firewall rules cleared";
   ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                            caller()->ice_connection_state(), kDefaultTimeout,
                            fake_clock);
   ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                            caller()->standardized_ice_connection_state(),
                            kDefaultTimeout, fake_clock);

   // According to RFC7675, if there is no response within 30 seconds then the
   // peer should consider the other side to have rejected the connection. This
   // is signaled by the state transitioning to "failed".
   constexpr int kConsentTimeout = 30000;
   for (const auto& caller_address : CallerAddresses()) {
     firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
   }
   RTC_LOG(LS_INFO) << "Firewall rules applied again";
   ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
                            caller()->ice_connection_state(), kConsentTimeout,
                            fake_clock);
   ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
                            caller()->standardized_ice_connection_state(),
                            kConsentTimeout, fake_clock);

   // We need to manually close the peerconnections before the fake clock goes
   // out of scope, or we trigger a DCHECK in rtp_sender.cc when we briefly
   // return to using non-faked time.
   delete SetCallerPcWrapperAndReturnCurrent(nullptr);
   delete SetCalleePcWrapperAndReturnCurrent(nullptr);
 }

 // Tests that if the connection doesn't get set up properly we eventually reach
 // the "failed" iceConnectionState.
 TEST_P(PeerConnectionIntegrationIceStatesTest, IceStateSetupFailure) {
   rtc::ScopedFakeClock fake_clock;
   // Some things use a time of "0" as a special value, so we need to start out
   // the fake clock at a nonzero time.
   fake_clock.AdvanceTime(TimeDelta::seconds(1));

   // Block connections to/from the caller and wait for ICE to become
   // disconnected.
   for (const auto& caller_address : CallerAddresses()) {
     firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
   }

   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   SetPortAllocatorFlags();
   SetUpNetworkInterfaces();
   caller()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();

   // According to RFC7675, if there is no response within 30 seconds then the
   // peer should consider the other side to have rejected the connection. This
   // is signaled by the state transitioning to "failed".
   constexpr int kConsentTimeout = 30000;
   ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
                            caller()->standardized_ice_connection_state(),
                            kConsentTimeout, fake_clock);

   // We need to manually close the peerconnections before the fake clock goes
   // out of scope, or we trigger a DCHECK in rtp_sender.cc when we briefly
   // return to using non-faked time.
   delete SetCallerPcWrapperAndReturnCurrent(nullptr);
   delete SetCalleePcWrapperAndReturnCurrent(nullptr);
 }

 // Tests that the best connection is set to the appropriate IPv4/IPv6 connection
 // and that the statistics in the metric observers are updated correctly.
 TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   SetPortAllocatorFlags();
   SetUpNetworkInterfaces();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();

   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // TODO(bugs.webrtc.org/9456): Fix it.
   const int num_best_ipv4 = webrtc::metrics::NumEvents(
       "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
   const int num_best_ipv6 = webrtc::metrics::NumEvents(
       "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
   if (TestIPv6()) {
     // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
     // connection.
     EXPECT_EQ(0, num_best_ipv4);
     EXPECT_EQ(1, num_best_ipv6);
   } else {
     EXPECT_EQ(1, num_best_ipv4);
     EXPECT_EQ(0, num_best_ipv6);
   }

   EXPECT_EQ(0, webrtc::metrics::NumEvents(
                    "WebRTC.PeerConnection.CandidatePairType_UDP",
                    webrtc::kIceCandidatePairHostHost));
   EXPECT_EQ(1, webrtc::metrics::NumEvents(
                    "WebRTC.PeerConnection.CandidatePairType_UDP",
                    webrtc::kIceCandidatePairHostPublicHostPublic));
 }

 constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
                                       cricket::PORTALLOCATOR_DISABLE_STUN |
                                       cricket::PORTALLOCATOR_DISABLE_RELAY;
 constexpr uint32_t kFlagsIPv6NoStun =
     cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN |
     cricket::PORTALLOCATOR_ENABLE_IPV6 | cricket::PORTALLOCATOR_DISABLE_RELAY;
 constexpr uint32_t kFlagsIPv4Stun =
     cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY;

 INSTANTIATE_TEST_SUITE_P(
     PeerConnectionIntegrationTest,
     PeerConnectionIntegrationIceStatesTest,
     Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
             Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
                    std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
                    std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));

 // This test sets up a call between two parties with audio and video.
 // During the call, the caller restarts ICE and the test verifies that
 // new ICE candidates are generated and audio and video still can flow, and the
 // ICE state reaches completed again.
 TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for ICE to complete.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);

   // To verify that the ICE restart actually occurs, get
   // ufrag/password/candidates before and after restart.
   // Create an SDP string of the first audio candidate for both clients.
   const webrtc::IceCandidateCollection* audio_candidates_caller =
       caller()->pc()->local_description()->candidates(0);
   const webrtc::IceCandidateCollection* audio_candidates_callee =
       callee()->pc()->local_description()->candidates(0);
   ASSERT_GT(audio_candidates_caller->count(), 0u);
   ASSERT_GT(audio_candidates_callee->count(), 0u);
   std::string caller_candidate_pre_restart;
   ASSERT_TRUE(
       audio_candidates_caller->at(0)->ToString(&caller_candidate_pre_restart));
   std::string callee_candidate_pre_restart;
   ASSERT_TRUE(
       audio_candidates_callee->at(0)->ToString(&callee_candidate_pre_restart));
   const cricket::SessionDescription* desc =
       caller()->pc()->local_description()->description();
   std::string caller_ufrag_pre_restart =
       desc->transport_infos()[0].description.ice_ufrag;
   desc = callee()->pc()->local_description()->description();
   std::string callee_ufrag_pre_restart =
       desc->transport_infos()[0].description.ice_ufrag;

   // Have the caller initiate an ICE restart.
   caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);

   // Grab the ufrags/candidates again.
   audio_candidates_caller = caller()->pc()->local_description()->candidates(0);
   audio_candidates_callee = callee()->pc()->local_description()->candidates(0);
   ASSERT_GT(audio_candidates_caller->count(), 0u);
   ASSERT_GT(audio_candidates_callee->count(), 0u);
   std::string caller_candidate_post_restart;
   ASSERT_TRUE(
       audio_candidates_caller->at(0)->ToString(&caller_candidate_post_restart));
   std::string callee_candidate_post_restart;
   ASSERT_TRUE(
       audio_candidates_callee->at(0)->ToString(&callee_candidate_post_restart));
   desc = caller()->pc()->local_description()->description();
   std::string caller_ufrag_post_restart =
       desc->transport_infos()[0].description.ice_ufrag;
   desc = callee()->pc()->local_description()->description();
   std::string callee_ufrag_post_restart =
       desc->transport_infos()[0].description.ice_ufrag;
   // Sanity check that an ICE restart was actually negotiated in SDP.
   ASSERT_NE(caller_candidate_pre_restart, caller_candidate_post_restart);
   ASSERT_NE(callee_candidate_pre_restart, callee_candidate_post_restart);
   ASSERT_NE(caller_ufrag_pre_restart, caller_ufrag_post_restart);
   ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart);

   // Ensure that additional frames are received after the ICE restart.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Verify that audio/video can be received end-to-end when ICE renomination is
 // enabled.
 TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
   PeerConnectionInterface::RTCConfiguration config;
   config.enable_ice_renomination = true;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for some frames to be received in each
   // direction.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Sanity check that ICE renomination was actually negotiated.
   const cricket::SessionDescription* desc =
       caller()->pc()->local_description()->description();
   for (const cricket::TransportInfo& info : desc->transport_infos()) {
     ASSERT_THAT(info.description.transport_options, Contains("renomination"));
   }
   desc = callee()->pc()->local_description()->description();
   for (const cricket::TransportInfo& info : desc->transport_infos()) {
     ASSERT_THAT(info.description.transport_options, Contains("renomination"));
   }
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // With a max bundle policy and RTCP muxing, adding a new media description to
 // the connection should not affect ICE at all because the new media will use
 // the existing connection.
 TEST_P(PeerConnectionIntegrationTest,
        AddMediaToConnectedBundleDoesNotRestartIce) {
   PeerConnectionInterface::RTCConfiguration config;
   config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
   config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(
       config, PeerConnectionInterface::RTCConfiguration()));
   ConnectFakeSignaling();

   caller()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kDefaultTimeout);

   caller()->clear_ice_connection_state_history();

   caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   EXPECT_EQ(0u, caller()->ice_connection_state_history().size());
 }

 // This test sets up a call between two parties with audio and video. It then
 // renegotiates setting the video m-line to "port 0", then later renegotiates
 // again, enabling video.
 TEST_P(PeerConnectionIntegrationTest,
        VideoFlowsAfterMediaSectionIsRejectedAndRecycled) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Do initial negotiation, only sending media from the caller. Will result in
   // video and audio recvonly "m=" sections.
   caller()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Negotiate again, disabling the video "m=" section (the callee will set the
   // port to 0 due to offer_to_receive_video = 0).
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_video = 0;
     callee()->SetOfferAnswerOptions(options);
   } else {
     callee()->SetRemoteOfferHandler([this] {
       callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
     });
   }
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Sanity check that video "m=" section was actually rejected.
   const ContentInfo* answer_video_content = cricket::GetFirstVideoContent(
       callee()->pc()->local_description()->description());
   ASSERT_NE(nullptr, answer_video_content);
   ASSERT_TRUE(answer_video_content->rejected);

   // Enable video and do negotiation again, making sure video is received
   // end-to-end, also adding media stream to callee.
   if (sdp_semantics_ == SdpSemantics::kPlanB) {
     PeerConnectionInterface::RTCOfferAnswerOptions options;
     options.offer_to_receive_video = 1;
     callee()->SetOfferAnswerOptions(options);
   } else {
     // The caller's transceiver is stopped, so we need to add another track.
     auto caller_transceiver =
         caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
     EXPECT_TRUE(caller_transceiver->stopped());
     caller()->AddVideoTrack();
   }
   callee()->AddVideoTrack();
   callee()->SetRemoteOfferHandler(nullptr);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Verify the caller receives frames from the newly added stream, and the
   // callee receives additional frames from the re-enabled video m= section.
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudio();
   media_expectations.ExpectBidirectionalVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This tests that if we negotiate after calling CreateSender but before we
 // have a track, then set a track later, frames from the newly-set track are
 // received end-to-end.
 TEST_F(PeerConnectionIntegrationTestPlanB,
        MediaFlowsAfterEarlyWarmupWithCreateSender) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   auto caller_audio_sender =
       caller()->pc()->CreateSender("audio", "caller_stream");
   auto caller_video_sender =
       caller()->pc()->CreateSender("video", "caller_stream");
   auto callee_audio_sender =
       callee()->pc()->CreateSender("audio", "callee_stream");
   auto callee_video_sender =
       callee()->pc()->CreateSender("video", "callee_stream");
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
   // Wait for ICE to complete, without any tracks being set.
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);
   // Now set the tracks, and expect frames to immediately start flowing.
   EXPECT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
   EXPECT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
   EXPECT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
   EXPECT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This tests that if we negotiate after calling AddTransceiver but before we
 // have a track, then set a track later, frames from the newly-set tracks are
 // received end-to-end.
 TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
        MediaFlowsAfterEarlyWarmupWithAddTransceiver) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   auto audio_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
   ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type());
   auto caller_audio_sender = audio_result.MoveValue()->sender();
   auto video_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
   ASSERT_EQ(RTCErrorType::NONE, video_result.error().type());
   auto caller_video_sender = video_result.MoveValue()->sender();
   callee()->SetRemoteOfferHandler([this] {
     ASSERT_EQ(2u, callee()->pc()->GetTransceivers().size());
     callee()->pc()->GetTransceivers()[0]->SetDirection(
         RtpTransceiverDirection::kSendRecv);
     callee()->pc()->GetTransceivers()[1]->SetDirection(
         RtpTransceiverDirection::kSendRecv);
   });
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
   // Wait for ICE to complete, without any tracks being set.
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);
   // Now set the tracks, and expect frames to immediately start flowing.
   auto callee_audio_sender = callee()->pc()->GetSenders()[0];
   auto callee_video_sender = callee()->pc()->GetSenders()[1];
   ASSERT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
   ASSERT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
   ASSERT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
   ASSERT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // This test verifies that a remote video track can be added via AddStream,
 // and sent end-to-end. For this particular test, it's simply echoed back
 // from the caller to the callee, rather than being forwarded to a third
 // PeerConnection.
 TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Just send a video track from the caller.
   caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
   ASSERT_EQ(1U, callee()->remote_streams()->count());

   // Echo the stream back, and do a new offer/anwer (initiated by callee this
   // time).
   callee()->pc()->AddStream(callee()->remote_streams()->at(0));
   callee()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);

   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test that we achieve the expected end-to-end connection time, using a
 // fake clock and simulated latency on the media and signaling paths.
 // We use a TURN<->TURN connection because this is usually the quickest to
 // set up initially, especially when we're confident the connection will work
 // and can start sending media before we get a STUN response.
 //
 // With various optimizations enabled, here are the network delays we expect to
 // be on the critical path:
 // 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then
 //                       signaling answer (with DTLS fingerprint).
 // 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
 //                  using TURN<->TURN pair, and DTLS exchange is 4 packets,
 //                  the first of which should have arrived before the answer.
 TEST_P(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
   rtc::ScopedFakeClock fake_clock;
   // Some things use a time of "0" as a special value, so we need to start out
   // the fake clock at a nonzero time.
   // TODO(deadbeef): Fix this.
   fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));

   static constexpr int media_hop_delay_ms = 50;
   static constexpr int signaling_trip_delay_ms = 500;
   // For explanation of these values, see comment above.
   static constexpr int required_media_hops = 9;
   static constexpr int required_signaling_trips = 2;
   // For internal delays (such as posting an event asychronously).
   static constexpr int allowed_internal_delay_ms = 20;
   static constexpr int total_connection_time_ms =
       media_hop_delay_ms * required_media_hops +
       signaling_trip_delay_ms * required_signaling_trips +
       allowed_internal_delay_ms;

   static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
                                                                  3478};
   static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
                                                                  0};
   static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
                                                                  3478};
   static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
                                                                  0};
   cricket::TestTurnServer* turn_server_1 = CreateTurnServer(
       turn_server_1_internal_address, turn_server_1_external_address);

   cricket::TestTurnServer* turn_server_2 = CreateTurnServer(
       turn_server_2_internal_address, turn_server_2_external_address);
   // Bypass permission check on received packets so media can be sent before
   // the candidate is signaled.
   network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_1] {
     turn_server_1->set_enable_permission_checks(false);
   });
   network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_2] {
     turn_server_2->set_enable_permission_checks(false);
   });

   PeerConnectionInterface::RTCConfiguration client_1_config;
   webrtc::PeerConnectionInterface::IceServer ice_server_1;
   ice_server_1.urls.push_back("turn:88.88.88.0:3478");
   ice_server_1.username = "test";
   ice_server_1.password = "test";
   client_1_config.servers.push_back(ice_server_1);
   client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
   client_1_config.presume_writable_when_fully_relayed = true;

   PeerConnectionInterface::RTCConfiguration client_2_config;
   webrtc::PeerConnectionInterface::IceServer ice_server_2;
   ice_server_2.urls.push_back("turn:99.99.99.0:3478");
   ice_server_2.username = "test";
   ice_server_2.password = "test";
   client_2_config.servers.push_back(ice_server_2);
   client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
   client_2_config.presume_writable_when_fully_relayed = true;

   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
   // Set up the simulated delays.
   SetSignalingDelayMs(signaling_trip_delay_ms);
   ConnectFakeSignaling();
   virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
   virtual_socket_server()->UpdateDelayDistribution();

   // Set "offer to receive audio/video" without adding any tracks, so we just
   // set up ICE/DTLS with no media.
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio = 1;
   options.offer_to_receive_video = 1;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
   EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms,
                              fake_clock);
   // Closing the PeerConnections destroys the ports before the ScopedFakeClock.
   // If this is not done a DCHECK can be hit in ports.cc, because a large
   // negative number is calculated for the rtt due to the global clock changing.
   ClosePeerConnections();
 }

 // Verify that a TurnCustomizer passed in through RTCConfiguration
 // is actually used by the underlying TURN candidate pair.
 // Note that turnport_unittest.cc contains more detailed, lower-level tests.
 TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
   static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
                                                                  3478};
   static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
                                                                  0};
   static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
                                                                  3478};
   static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
                                                                  0};
   CreateTurnServer(turn_server_1_internal_address,
                    turn_server_1_external_address);
   CreateTurnServer(turn_server_2_internal_address,
                    turn_server_2_external_address);

   PeerConnectionInterface::RTCConfiguration client_1_config;
   webrtc::PeerConnectionInterface::IceServer ice_server_1;
   ice_server_1.urls.push_back("turn:88.88.88.0:3478");
   ice_server_1.username = "test";
   ice_server_1.password = "test";
   client_1_config.servers.push_back(ice_server_1);
   client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
   auto* customizer1 = CreateTurnCustomizer();
   client_1_config.turn_customizer = customizer1;

   PeerConnectionInterface::RTCConfiguration client_2_config;
   webrtc::PeerConnectionInterface::IceServer ice_server_2;
   ice_server_2.urls.push_back("turn:99.99.99.0:3478");
   ice_server_2.username = "test";
   ice_server_2.password = "test";
   client_2_config.servers.push_back(ice_server_2);
   client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
   auto* customizer2 = CreateTurnCustomizer();
   client_2_config.turn_customizer = customizer2;

   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
   ConnectFakeSignaling();

   // Set "offer to receive audio/video" without adding any tracks, so we just
   // set up ICE/DTLS with no media.
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio = 1;
   options.offer_to_receive_video = 1;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);

   ExpectTurnCustomizerCountersIncremented(customizer1);
   ExpectTurnCustomizerCountersIncremented(customizer2);
 }

 // Verifies that you can use TCP instead of UDP to connect to a TURN server and
 // send media between the caller and the callee.
 TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
   static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                3478};
   static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

   // Enable TCP for the fake turn server.
   CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                    cricket::PROTO_TCP);

   webrtc::PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
   ice_server.username = "test";
   ice_server.password = "test";

   PeerConnectionInterface::RTCConfiguration client_1_config;
   client_1_config.servers.push_back(ice_server);
   client_1_config.type = webrtc::PeerConnectionInterface::kRelay;

   PeerConnectionInterface::RTCConfiguration client_2_config;
   client_2_config.servers.push_back(ice_server);
   client_2_config.type = webrtc::PeerConnectionInterface::kRelay;

   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));

   // Do normal offer/answer and wait for ICE to complete.
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);

   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   EXPECT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Verify that a SSLCertificateVerifier passed in through
 // PeerConnectionDependencies is actually used by the underlying SSL
 // implementation to determine whether a certificate presented by the TURN
 // server is accepted by the client. Note that openssladapter_unittest.cc
 // contains more detailed, lower-level tests.
 TEST_P(PeerConnectionIntegrationTest,
        SSLCertificateVerifierUsedForTurnConnections) {
   static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                3478};
   static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

   // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
   // that host name verification passes on the fake certificate.
   CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                    cricket::PROTO_TLS, "88.88.88.0");

   webrtc::PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
   ice_server.username = "test";
   ice_server.password = "test";

   PeerConnectionInterface::RTCConfiguration client_1_config;
   client_1_config.servers.push_back(ice_server);
   client_1_config.type = webrtc::PeerConnectionInterface::kRelay;

   PeerConnectionInterface::RTCConfiguration client_2_config;
   client_2_config.servers.push_back(ice_server);
   // Setting the type to kRelay forces the connection to go through a TURN
   // server.
   client_2_config.type = webrtc::PeerConnectionInterface::kRelay;

   // Get a copy to the pointer so we can verify calls later.
   rtc::TestCertificateVerifier* client_1_cert_verifier =
       new rtc::TestCertificateVerifier();
   client_1_cert_verifier->verify_certificate_ = true;
   rtc::TestCertificateVerifier* client_2_cert_verifier =
       new rtc::TestCertificateVerifier();
   client_2_cert_verifier->verify_certificate_ = true;

   // Create the dependencies with the test certificate verifier.
   webrtc::PeerConnectionDependencies client_1_deps(nullptr);
   client_1_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
   webrtc::PeerConnectionDependencies client_2_deps(nullptr);
   client_2_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);

   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
       client_1_config, std::move(client_1_deps), client_2_config,
       std::move(client_2_deps)));
   ConnectFakeSignaling();

   // Set "offer to receive audio/video" without adding any tracks, so we just
   // set up ICE/DTLS with no media.
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio = 1;
   options.offer_to_receive_video = 1;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);

   EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
   EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
 }

 TEST_P(PeerConnectionIntegrationTest,
        SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection) {
   static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                3478};
   static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

   // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
   // that host name verification passes on the fake certificate.
   CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                    cricket::PROTO_TLS, "88.88.88.0");

   webrtc::PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
   ice_server.username = "test";
   ice_server.password = "test";

   PeerConnectionInterface::RTCConfiguration client_1_config;
   client_1_config.servers.push_back(ice_server);
   client_1_config.type = webrtc::PeerConnectionInterface::kRelay;

   PeerConnectionInterface::RTCConfiguration client_2_config;
   client_2_config.servers.push_back(ice_server);
   // Setting the type to kRelay forces the connection to go through a TURN
   // server.
   client_2_config.type = webrtc::PeerConnectionInterface::kRelay;

   // Get a copy to the pointer so we can verify calls later.
   rtc::TestCertificateVerifier* client_1_cert_verifier =
       new rtc::TestCertificateVerifier();
   client_1_cert_verifier->verify_certificate_ = false;
   rtc::TestCertificateVerifier* client_2_cert_verifier =
       new rtc::TestCertificateVerifier();
   client_2_cert_verifier->verify_certificate_ = false;

   // Create the dependencies with the test certificate verifier.
   webrtc::PeerConnectionDependencies client_1_deps(nullptr);
   client_1_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
   webrtc::PeerConnectionDependencies client_2_deps(nullptr);
   client_2_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);

   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
       client_1_config, std::move(client_1_deps), client_2_config,
       std::move(client_2_deps)));
   ConnectFakeSignaling();

   // Set "offer to receive audio/video" without adding any tracks, so we just
   // set up ICE/DTLS with no media.
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio = 1;
   options.offer_to_receive_video = 1;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
   bool wait_res = true;
   // TODO(bugs.webrtc.org/9219): When IceConnectionState is implemented
   // properly, should be able to just wait for a state of "failed" instead of
   // waiting a fixed 10 seconds.
   WAIT_(DtlsConnected(), kDefaultTimeout, wait_res);
   ASSERT_FALSE(wait_res);

   EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
   EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
 }

 // Test that audio and video flow end-to-end when codec names don't use the
 // expected casing, given that they're supposed to be case insensitive. To test
 // this, all but one codec is removed from each media description, and its
 // casing is changed.
 //
 // In the past, this has regressed and caused crashes/black video, due to the
 // fact that code at some layers was doing case-insensitive comparisons and
 // code at other layers was not.
 TEST_P(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();

   // Remove all but one audio/video codec (opus and VP8), and change the
   // casing of the caller's generated offer.
   caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) {
     cricket::AudioContentDescription* audio =
         GetFirstAudioContentDescription(description);
     ASSERT_NE(nullptr, audio);
     auto audio_codecs = audio->codecs();
     audio_codecs.erase(std::remove_if(audio_codecs.begin(), audio_codecs.end(),
                                       [](const cricket::AudioCodec& codec) {
                                         return codec.name != "opus";
                                       }),
                        audio_codecs.end());
     ASSERT_EQ(1u, audio_codecs.size());
     audio_codecs[0].name = "OpUs";
     audio->set_codecs(audio_codecs);

     cricket::VideoContentDescription* video =
         GetFirstVideoContentDescription(description);
     ASSERT_NE(nullptr, video);
     auto video_codecs = video->codecs();
     video_codecs.erase(std::remove_if(video_codecs.begin(), video_codecs.end(),
                                       [](const cricket::VideoCodec& codec) {
                                         return codec.name != "VP8";
                                       }),
                        video_codecs.end());
     ASSERT_EQ(1u, video_codecs.size());
     video_codecs[0].name = "vP8";
     video->set_codecs(video_codecs);
   });

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Verify frames are still received end-to-end.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for one audio frame to be received by the callee.
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudio(1);
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
   ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
   auto receiver = callee()->pc()->GetReceivers()[0];
   ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
   auto sources = receiver->GetSources();
   ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
   EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
             sources[0].source_id());
   EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
 }

 TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for one video frame to be received by the callee.
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeVideo(1);
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
   ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
   auto receiver = callee()->pc()->GetReceivers()[0];
   ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_VIDEO);
   auto sources = receiver->GetSources();
   ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
   EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
             sources[0].source_id());
   EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
 }

 // Test that if a track is removed and added again with a different stream ID,
 // the new stream ID is successfully communicated in SDP and media continues to
 // flow end-to-end.
 // TODO(webrtc.bugs.org/8734): This test does not work for Unified Plan because
 // it will not reuse a transceiver that has already been sending. After creating
 // a new transceiver it tries to create an offer with two senders of the same
 // track ids and it fails.
 TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Add track using stream 1, do offer/answer.
   rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
       caller()->CreateLocalAudioTrack();
   rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
       caller()->AddTrack(track, {"stream_1"});
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   {
     MediaExpectations media_expectations;
     media_expectations.CalleeExpectsSomeAudio(1);
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
   // Remove the sender, and create a new one with the new stream.
   caller()->pc()->RemoveTrack(sender);
   sender = caller()->AddTrack(track, {"stream_2"});
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for additional audio frames to be received by the callee.
   {
     MediaExpectations media_expectations;
     media_expectations.CalleeExpectsSomeAudio();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
 }

 TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   auto output = absl::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
   ON_CALL(*output, IsActive()).WillByDefault(::testing::Return(true));
   ON_CALL(*output, Write(::testing::_)).WillByDefault(::testing::Return(true));
   EXPECT_CALL(*output, Write(::testing::_)).Times(::testing::AtLeast(1));
   EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
       std::move(output), webrtc::RtcEventLog::kImmediateOutput));

   caller()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 }

 // Test that if candidates are only signaled by applying full session
 // descriptions (instead of using AddIceCandidate), the peers can connect to
 // each other and exchange media.
 TEST_P(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   // Each side will signal the session descriptions but not candidates.
   ConnectFakeSignalingForSdpOnly();

   // Add audio video track and exchange the initial offer/answer with media
   // information only. This will start ICE gathering on each side.
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();

   // Wait for all candidates to be gathered on both the caller and callee.
   ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                  caller()->ice_gathering_state(), kDefaultTimeout);
   ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                  callee()->ice_gathering_state(), kDefaultTimeout);

   // The candidates will now be included in the session description, so
   // signaling them will start the ICE connection.
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Ensure that media flows in both directions.
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test that SetAudioPlayout can be used to disable audio playout from the
 // start, then later enable it. This may be useful, for example, if the caller
 // needs to play a local ringtone until some event occurs, after which it
 // switches to playing the received audio.
 TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Set up audio-only call where audio playout is disabled on caller's side.
   caller()->pc()->SetAudioPlayout(false);
   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Pump messages for a second.
   WAIT(false, 1000);
   // Since audio playout is disabled, the caller shouldn't have received
   // anything (at the playout level, at least).
   EXPECT_EQ(0, caller()->audio_frames_received());
   // As a sanity check, make sure the callee (for which playout isn't disabled)
   // did still see frames on its audio level.
   ASSERT_GT(callee()->audio_frames_received(), 0);

   // Enable playout again, and ensure audio starts flowing.
   caller()->pc()->SetAudioPlayout(true);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudio();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 double GetAudioEnergyStat(PeerConnectionWrapper* pc) {
   auto report = pc->NewGetStats();
   auto track_stats_list =
       report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>();
   const webrtc::RTCMediaStreamTrackStats* remote_track_stats = nullptr;
   for (const auto* track_stats : track_stats_list) {
     if (track_stats->remote_source.is_defined() &&
         *track_stats->remote_source) {
       remote_track_stats = track_stats;
       break;
     }
   }

   if (!remote_track_stats->total_audio_energy.is_defined()) {
     return 0.0;
   }
   return *remote_track_stats->total_audio_energy;
 }

 // Test that if audio playout is disabled via the SetAudioPlayout() method, then
 // incoming audio is still processed and statistics are generated.
 TEST_P(PeerConnectionIntegrationTest,
        DisableAudioPlayoutStillGeneratesAudioStats) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Set up audio-only call where playout is disabled but audio-processing is
   // still active.
   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   caller()->pc()->SetAudioPlayout(false);

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Wait for the callee to receive audio stats.
   EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs);
 }

 // Test that SetAudioRecording can be used to disable audio recording from the
 // start, then later enable it. This may be useful, for example, if the caller
 // wants to ensure that no audio resources are active before a certain state
 // is reached.
 TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();

   // Set up audio-only call where audio recording is disabled on caller's side.
   caller()->pc()->SetAudioRecording(false);
   caller()->AddAudioTrack();
   callee()->AddAudioTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Pump messages for a second.
   WAIT(false, 1000);
   // Since caller has disabled audio recording, the callee shouldn't have
   // received anything.
   EXPECT_EQ(0, callee()->audio_frames_received());
   // As a sanity check, make sure the caller did still see frames on its
   // audio level since audio recording is enabled on the calle side.
   ASSERT_GT(caller()->audio_frames_received(), 0);

   // Enable audio recording again, and ensure audio starts flowing.
   caller()->pc()->SetAudioRecording(true);
   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudio();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test that after closing PeerConnections, they stop sending any packets (ICE,
 // DTLS, RTP...).
 TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
   // Set up audio/video/data, wait for some frames to be received.
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
 #ifdef HAVE_SCTP
   caller()->CreateDataChannel();
 #endif
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
   // Close PeerConnections.
   ClosePeerConnections();
   // Pump messages for a second, and ensure no new packets end up sent.
   uint32_t sent_packets_a = virtual_socket_server()->sent_packets();
   WAIT(false, 1000);
   uint32_t sent_packets_b = virtual_socket_server()->sent_packets();
   EXPECT_EQ(sent_packets_a, sent_packets_b);
 }

 // Test that transport stats are generated by the RTCStatsCollector for a
 // connection that only involves data channels. This is a regression test for
 // crbug.com/826972.
 #ifdef HAVE_SCTP
 TEST_P(PeerConnectionIntegrationTest,
        TransportStatsReportedForDataChannelOnlyConnection) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->CreateDataChannel();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);

   auto caller_report = caller()->NewGetStats();
   EXPECT_EQ(1u, caller_report->GetStatsOfType<RTCTransportStats>().size());
   auto callee_report = callee()->NewGetStats();
   EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
 }
 #endif  // HAVE_SCTP

 TEST_P(PeerConnectionIntegrationTest,
        IceEventsGeneratedAndLoggedInRtcEventLog) {
   ASSERT_TRUE(CreatePeerConnectionWrappersWithFakeRtcEventLog());
   ConnectFakeSignaling();
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio = 1;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
   ASSERT_NE(nullptr, caller()->event_log_factory());
   ASSERT_NE(nullptr, callee()->event_log_factory());
   webrtc::FakeRtcEventLog* caller_event_log =
       static_cast<webrtc::FakeRtcEventLog*>(
           caller()->event_log_factory()->last_log_created());
   webrtc::FakeRtcEventLog* callee_event_log =
       static_cast<webrtc::FakeRtcEventLog*>(
           callee()->event_log_factory()->last_log_created());
   ASSERT_NE(nullptr, caller_event_log);
   ASSERT_NE(nullptr, callee_event_log);
   int caller_ice_config_count = caller_event_log->GetEventCount(
       webrtc::RtcEvent::Type::IceCandidatePairConfig);
   int caller_ice_event_count = caller_event_log->GetEventCount(
       webrtc::RtcEvent::Type::IceCandidatePairEvent);
   int callee_ice_config_count = callee_event_log->GetEventCount(
       webrtc::RtcEvent::Type::IceCandidatePairConfig);
   int callee_ice_event_count = callee_event_log->GetEventCount(
       webrtc::RtcEvent::Type::IceCandidatePairEvent);
   EXPECT_LT(0, caller_ice_config_count);
   EXPECT_LT(0, caller_ice_event_count);
   EXPECT_LT(0, callee_ice_config_count);
   EXPECT_LT(0, callee_ice_event_count);
 }

 TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
   static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                3478};
   static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

   CreateTurnServer(turn_server_internal_address, turn_server_external_address);

   webrtc::PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turn:88.88.88.0:3478");
   ice_server.username = "test";
   ice_server.password = "test";

   PeerConnectionInterface::RTCConfiguration caller_config;
   caller_config.servers.push_back(ice_server);
   caller_config.type = webrtc::PeerConnectionInterface::kRelay;
   caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
   caller_config.surface_ice_candidates_on_ice_transport_type_changed = true;

   PeerConnectionInterface::RTCConfiguration callee_config;
   callee_config.servers.push_back(ice_server);
   callee_config.type = webrtc::PeerConnectionInterface::kRelay;
   callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
   callee_config.surface_ice_candidates_on_ice_transport_type_changed = true;

   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));

   // Do normal offer/answer and wait for ICE to complete.
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Since we are doing continual gathering, the ICE transport does not reach
   // kIceGatheringComplete (see
   // P2PTransportChannel::OnCandidatesAllocationDone), and consequently not
   // kIceConnectionComplete.
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  caller()->ice_connection_state(), kDefaultTimeout);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);
   // Note that we cannot use the metric
   // |WebRTC.PeerConnection.CandidatePairType_UDP| in this test since this
   // metric is only populated when we reach kIceConnectionComplete in the
   // current implementation.
   EXPECT_EQ(cricket::RELAY_PORT_TYPE,
             caller()->last_candidate_gathered().type());
   EXPECT_EQ(cricket::RELAY_PORT_TYPE,
             callee()->last_candidate_gathered().type());

   // Loosen the caller's candidate filter.
   caller_config = caller()->pc()->GetConfiguration();
   caller_config.type = webrtc::PeerConnectionInterface::kAll;
   caller()->pc()->SetConfiguration(caller_config);
   // We should have gathered a new host candidate.
   EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
                  caller()->last_candidate_gathered().type(), kDefaultTimeout);

   // Loosen the callee's candidate filter.
   callee_config = callee()->pc()->GetConfiguration();
   callee_config.type = webrtc::PeerConnectionInterface::kAll;
   callee()->pc()->SetConfiguration(callee_config);
   EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
                  callee()->last_candidate_gathered().type(), kDefaultTimeout);
 }

 TEST_P(PeerConnectionIntegrationTest, OnIceCandidateError) {
   static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                                3478};
   static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

   CreateTurnServer(turn_server_internal_address, turn_server_external_address);

   webrtc::PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turn:88.88.88.0:3478");
   ice_server.username = "test";
   ice_server.password = "123";

   PeerConnectionInterface::RTCConfiguration caller_config;
   caller_config.servers.push_back(ice_server);
   caller_config.type = webrtc::PeerConnectionInterface::kRelay;
   caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;

   PeerConnectionInterface::RTCConfiguration callee_config;
   callee_config.servers.push_back(ice_server);
   callee_config.type = webrtc::PeerConnectionInterface::kRelay;
   callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;

   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));

   // Do normal offer/answer and wait for ICE to complete.
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   EXPECT_EQ_WAIT(401, caller()->error_event().error_code, kDefaultTimeout);
   EXPECT_EQ("Unauthorized", caller()->error_event().error_text);
   EXPECT_EQ("turn:88.88.88.0:3478?transport=udp", caller()->error_event().url);
   EXPECT_NE(std::string::npos,
             caller()->error_event().host_candidate.find(":"));
 }

 INSTANTIATE_TEST_SUITE_P(PeerConnectionIntegrationTest,
                          PeerConnectionIntegrationTest,
                          Values(SdpSemantics::kPlanB,
                                 SdpSemantics::kUnifiedPlan));

 // Tests that verify interoperability between Plan B and Unified Plan
 // PeerConnections.
 class PeerConnectionIntegrationInteropTest
     : public PeerConnectionIntegrationBaseTest,
       public ::testing::WithParamInterface<
           std::tuple<SdpSemantics, SdpSemantics>> {
  protected:
   // Setting the SdpSemantics for the base test to kDefault does not matter
   // because we specify not to use the test semantics when creating
   // PeerConnectionWrappers.
   PeerConnectionIntegrationInteropTest()
       : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB),
         caller_semantics_(std::get<0>(GetParam())),
         callee_semantics_(std::get<1>(GetParam())) {}

   bool CreatePeerConnectionWrappersWithSemantics() {
     return CreatePeerConnectionWrappersWithSdpSemantics(caller_semantics_,
                                                         callee_semantics_);
   }

   const SdpSemantics caller_semantics_;
   const SdpSemantics callee_semantics_;
 };

 TEST_P(PeerConnectionIntegrationInteropTest, NoMediaLocalToNoMediaRemote) {
   ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
   ConnectFakeSignaling();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 }

 TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) {
   ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
   ConnectFakeSignaling();
   auto audio_sender = caller()->AddAudioTrack();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Verify that one audio receiver has been created on the remote and that it
   // has the same track ID as the sending track.
   auto receivers = callee()->pc()->GetReceivers();
   ASSERT_EQ(1u, receivers.size());
   EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type());
   EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id());

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudio();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) {
   ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
   ConnectFakeSignaling();
   auto video_sender = caller()->AddVideoTrack();
   auto audio_sender = caller()->AddAudioTrack();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Verify that one audio and one video receiver have been created on the
   // remote and that they have the same track IDs as the sending tracks.
   auto audio_receivers =
       callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO);
   ASSERT_EQ(1u, audio_receivers.size());
   EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id());
   auto video_receivers =
       callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO);
   ASSERT_EQ(1u, video_receivers.size());
   EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id());

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 TEST_P(PeerConnectionIntegrationInteropTest,
        OneAudioOneVideoLocalToOneAudioOneVideoRemote) {
   ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   MediaExpectations media_expectations;
   media_expectations.ExpectBidirectionalAudioAndVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 TEST_P(PeerConnectionIntegrationInteropTest,
        ReverseRolesOneAudioLocalToOneVideoRemote) {
   ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
   callee()->AddVideoTrack();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Verify that only the audio track has been negotiated.
   EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size());
   // Might also check that the callee's NegotiationNeeded flag is set.

   // Reverse roles.
   callee()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   MediaExpectations media_expectations;
   media_expectations.CallerExpectsSomeVideo();
   media_expectations.CalleeExpectsSomeAudio();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 INSTANTIATE_TEST_SUITE_P(
     PeerConnectionIntegrationTest,
     PeerConnectionIntegrationInteropTest,
     Values(std::make_tuple(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
            std::make_tuple(SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB)));

 // Test that if the Unified Plan side offers two video tracks then the Plan B
 // side will only see the first one and ignore the second.
 TEST_F(PeerConnectionIntegrationTestPlanB, TwoVideoUnifiedPlanToNoMediaPlanB) {
   ASSERT_TRUE(CreatePeerConnectionWrappersWithSdpSemantics(
       SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB));
   ConnectFakeSignaling();
   auto first_sender = caller()->AddVideoTrack();
   caller()->AddVideoTrack();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

   // Verify that there is only one receiver and it corresponds to the first
   // added track.
   auto receivers = callee()->pc()->GetReceivers();
   ASSERT_EQ(1u, receivers.size());
   EXPECT_TRUE(receivers[0]->track()->enabled());
   EXPECT_EQ(first_sender->track()->id(), receivers[0]->track()->id());

   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeVideo();
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }

 // Test that if the initial offer tagged BUNDLE section is rejected due to its
 // associated RtpTransceiver being stopped and another transceiver is added,
 // then renegotiation causes the callee to receive the new video track without
 // error.
 // This is a regression test for bugs.webrtc.org/9954
 TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
        ReOfferWithStoppedBundleTaggedTransceiver) {
   RTCConfiguration config;
   config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
   ConnectFakeSignaling();
   auto audio_transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
   ASSERT_TRUE(audio_transceiver_or_error.ok());
   auto audio_transceiver = audio_transceiver_or_error.MoveValue();

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   {
     MediaExpectations media_expectations;
     media_expectations.CalleeExpectsSomeAudio();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }

   audio_transceiver->Stop();
   caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());

   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   {
     MediaExpectations media_expectations;
     media_expectations.CalleeExpectsSomeVideo();
     ASSERT_TRUE(ExpectNewFrames(media_expectations));
   }
 }

 #ifdef HAVE_SCTP

 TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
        EndToEndCallWithBundledSctpDataChannel) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->CreateDataChannel();
   caller()->AddAudioVideoTracks();
   callee()->AddAudioVideoTracks();
   caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Ensure that media and data are multiplexed on the same DTLS transport.
   // This only works on Unified Plan, because transports are not exposed in plan
   // B.
   auto sctp_info = caller()->pc()->GetSctpTransport()->Information();
   EXPECT_EQ(sctp_info.dtls_transport(),
             caller()->pc()->GetSenders()[0]->dtls_transport());
 }

 #endif  // HAVE_SCTP

 }  // namespace
 }  // namespace webrtc

 #endif  // if !defined(THREAD_SANITIZER)